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40
.claude/skills/changelog/SKILL.md
Normal file
40
.claude/skills/changelog/SKILL.md
Normal file
@@ -0,0 +1,40 @@
|
||||
---
|
||||
name: changelog
|
||||
description: Create changelog files for important commits in a PR
|
||||
---
|
||||
|
||||
Create changelog files for the important commits in this PR. The PR number is provided as an argument.
|
||||
|
||||
## Instructions
|
||||
|
||||
1. First, check what commits are on the current branch compared to main:
|
||||
```
|
||||
git log main..HEAD --oneline
|
||||
```
|
||||
|
||||
2. For each significant change, create a changelog file in the `changelog/` folder using the format:
|
||||
- `{PR_NUMBER}.added.md` - for new features
|
||||
- `{PR_NUMBER}.added.2.md`, `{PR_NUMBER}.added.3.md` - for additional new features
|
||||
- `{PR_NUMBER}.changed.md` - for changes to existing functionality
|
||||
- `{PR_NUMBER}.fixed.md` - for bug fixes
|
||||
- `{PR_NUMBER}.deprecated.md` - for deprecations
|
||||
|
||||
3. Each changelog file should at least contain a main single line starting with `- ` followed by a clear description of the change.
|
||||
|
||||
4. If the change is complicated, changelog files can have indented lines after the main line with additional details or code samples.
|
||||
|
||||
5. Use ⚠️ emoji prefix for breaking changes.
|
||||
|
||||
## Example
|
||||
|
||||
For PR #3519 with a new feature and a bug fix:
|
||||
|
||||
`changelog/3519.added.md`:
|
||||
```
|
||||
- Added `SomeNewFeature` for doing something useful.
|
||||
```
|
||||
|
||||
`changelog/3519.fixed.md`:
|
||||
```
|
||||
- Fixed an issue where something was not working correctly.
|
||||
```
|
||||
257
.claude/skills/docstring/SKILL.md
Normal file
257
.claude/skills/docstring/SKILL.md
Normal file
@@ -0,0 +1,257 @@
|
||||
---
|
||||
name: docstring
|
||||
description: Document a Python module and its classes using Google style
|
||||
---
|
||||
|
||||
Document a Python module and its classes using Google-style docstrings following project conventions. The class name is provided as an argument.
|
||||
|
||||
## Instructions
|
||||
|
||||
1. First, find the class in the codebase:
|
||||
```
|
||||
Search for "class ClassName" in src/pipecat/
|
||||
```
|
||||
|
||||
2. If multiple files contain that class name:
|
||||
- List all matches with their file paths
|
||||
- Ask the user which one they want to document
|
||||
- Wait for confirmation before proceeding
|
||||
|
||||
3. Once the file is identified, read the module to understand its structure:
|
||||
- Identify all classes, functions, and important type aliases
|
||||
- Understand the purpose of each component
|
||||
|
||||
4. Apply documentation in this order:
|
||||
- Module docstring (at top, after imports)
|
||||
- Class docstrings
|
||||
- `__init__` methods (always document constructor parameters)
|
||||
- Public methods (not starting with `_`)
|
||||
- Dataclass/config classes with field descriptions
|
||||
|
||||
5. Skip documentation for:
|
||||
- Private methods (starting with `_`)
|
||||
- Simple dunder methods (`__str__`, `__repr__`, `__post_init__`)
|
||||
- Very simple pass-through properties
|
||||
- **Already documented code** - If a class, method, or function already has a complete docstring that follows the project style, do not modify it. A docstring is complete if it has:
|
||||
- A one-line summary
|
||||
- Args section (if it has parameters)
|
||||
- Returns section (if it returns something meaningful)
|
||||
- Only add or improve documentation where it is missing or incomplete
|
||||
|
||||
## Module Docstring Format
|
||||
|
||||
```python
|
||||
"""[One-line description of module purpose].
|
||||
|
||||
[Optional: Longer explanation of functionality, key classes, or use cases.]
|
||||
"""
|
||||
```
|
||||
|
||||
Example:
|
||||
```python
|
||||
"""Neuphonic text-to-speech service implementations.
|
||||
|
||||
This module provides WebSocket and HTTP-based integrations with Neuphonic's
|
||||
text-to-speech API for real-time audio synthesis.
|
||||
"""
|
||||
```
|
||||
|
||||
## Class Docstring Format
|
||||
|
||||
```python
|
||||
class ClassName:
|
||||
"""One-line summary describing what the class does.
|
||||
|
||||
[Longer description explaining purpose, behavior, and key features.
|
||||
Use action-oriented language.]
|
||||
|
||||
[Optional: Event handlers, usage notes, or important caveats.]
|
||||
"""
|
||||
```
|
||||
|
||||
Example:
|
||||
```python
|
||||
class FrameProcessor(BaseObject):
|
||||
"""Base class for all frame processors in the pipeline.
|
||||
|
||||
Frame processors are the building blocks of Pipecat pipelines, they can be
|
||||
linked to form complex processing pipelines. They receive frames, process
|
||||
them, and pass them to the next or previous processor in the chain.
|
||||
|
||||
Event handlers available:
|
||||
|
||||
- on_before_process_frame: Called before a frame is processed
|
||||
- on_after_process_frame: Called after a frame is processed
|
||||
|
||||
Example::
|
||||
|
||||
@processor.event_handler("on_before_process_frame")
|
||||
async def on_before_process_frame(processor, frame):
|
||||
...
|
||||
|
||||
@processor.event_handler("on_after_process_frame")
|
||||
async def on_after_process_frame(processor, frame):
|
||||
...
|
||||
"""
|
||||
```
|
||||
|
||||
Note: When listing event handlers, do NOT use backticks. Include an `Example::` section (with double colon for Sphinx) showing the decorator pattern and function signature for each event.
|
||||
|
||||
## Constructor (`__init__`) Format
|
||||
|
||||
```python
|
||||
def __init__(self, *, param1: Type, param2: Type = default, **kwargs):
|
||||
"""Initialize the [ClassName].
|
||||
|
||||
Args:
|
||||
param1: Description of param1 and its purpose.
|
||||
param2: Description of param2. Defaults to [default].
|
||||
**kwargs: Additional arguments passed to parent class.
|
||||
"""
|
||||
```
|
||||
|
||||
Example:
|
||||
```python
|
||||
def __init__(
|
||||
self,
|
||||
*,
|
||||
api_key: str,
|
||||
voice_id: Optional[str] = None,
|
||||
sample_rate: Optional[int] = 22050,
|
||||
**kwargs,
|
||||
):
|
||||
"""Initialize the Neuphonic TTS service.
|
||||
|
||||
Args:
|
||||
api_key: Neuphonic API key for authentication.
|
||||
voice_id: ID of the voice to use for synthesis.
|
||||
sample_rate: Audio sample rate in Hz. Defaults to 22050.
|
||||
**kwargs: Additional arguments passed to parent InterruptibleTTSService.
|
||||
"""
|
||||
```
|
||||
|
||||
## Method Docstring Format
|
||||
|
||||
```python
|
||||
async def method_name(self, param1: Type) -> ReturnType:
|
||||
"""One-line summary of what method does.
|
||||
|
||||
[Longer description if behavior isn't obvious.]
|
||||
|
||||
Args:
|
||||
param1: Description of param1.
|
||||
|
||||
Returns:
|
||||
Description of return value.
|
||||
|
||||
Raises:
|
||||
ExceptionType: When this exception is raised.
|
||||
"""
|
||||
```
|
||||
|
||||
Example:
|
||||
```python
|
||||
async def put(self, item: Tuple[Frame, FrameDirection, FrameCallback]):
|
||||
"""Put an item into the priority queue.
|
||||
|
||||
System frames (`SystemFrame`) have higher priority than any other
|
||||
frames. If a non-frame item is provided it will have the highest priority.
|
||||
|
||||
Args:
|
||||
item: The item to enqueue.
|
||||
"""
|
||||
```
|
||||
|
||||
## Dataclass/Config Format
|
||||
|
||||
```python
|
||||
@dataclass
|
||||
class ConfigName:
|
||||
"""One-line description of configuration.
|
||||
|
||||
[Explanation of when/how to use this config.]
|
||||
|
||||
Parameters:
|
||||
field1: Description of field1.
|
||||
field2: Description of field2. Defaults to [default].
|
||||
"""
|
||||
|
||||
field1: Type
|
||||
field2: Type = default_value
|
||||
```
|
||||
|
||||
Example:
|
||||
```python
|
||||
@dataclass
|
||||
class FrameProcessorSetup:
|
||||
"""Configuration parameters for frame processor initialization.
|
||||
|
||||
Parameters:
|
||||
clock: The clock instance for timing operations.
|
||||
task_manager: The task manager for handling async operations.
|
||||
observer: Optional observer for monitoring frame processing events.
|
||||
"""
|
||||
|
||||
clock: BaseClock
|
||||
task_manager: BaseTaskManager
|
||||
observer: Optional[BaseObserver] = None
|
||||
```
|
||||
|
||||
## Enum Documentation Format
|
||||
|
||||
```python
|
||||
class EnumName(Enum):
|
||||
"""One-line description of the enum purpose.
|
||||
|
||||
[Longer description of how the enum is used.]
|
||||
|
||||
Parameters:
|
||||
VALUE1: Description of VALUE1.
|
||||
VALUE2: Description of VALUE2.
|
||||
"""
|
||||
|
||||
VALUE1 = 1
|
||||
VALUE2 = 2
|
||||
```
|
||||
|
||||
## Writing Style Guidelines
|
||||
|
||||
- **Concise and professional** - No casual language or filler words
|
||||
- **Action-oriented** - Start with verbs: "Processes...", "Manages...", "Converts..."
|
||||
- **Purpose before implementation** - Explain WHY before HOW
|
||||
- **Clear parameter descriptions** - Include type hints, defaults, and purpose
|
||||
- **No redundant type info** - Type hints are in the signature, don't repeat in description
|
||||
- **Use backticks for code references** - Wrap class names, method names, event names, parameter names, and code snippets in backticks
|
||||
|
||||
Good: "Neuphonic API key for authentication."
|
||||
Bad: "str: The API key (string) that is used for authenticating with Neuphonic."
|
||||
|
||||
Good: "Triggers `on_speech_started` when the `VADAnalyzer` detects speech."
|
||||
Bad: "Triggers on_speech_started when the VADAnalyzer detects speech."
|
||||
|
||||
## Deprecation Notice Format
|
||||
|
||||
When documenting deprecated code:
|
||||
|
||||
```python
|
||||
"""[Description].
|
||||
|
||||
.. deprecated:: X.X.X
|
||||
`ClassName` is deprecated and will be removed in a future version.
|
||||
Use `NewClassName` instead.
|
||||
"""
|
||||
```
|
||||
|
||||
## Checklist
|
||||
|
||||
Before finishing, verify:
|
||||
|
||||
- [ ] Module has a docstring at the top (after copyright header and imports)
|
||||
- [ ] All public classes have docstrings
|
||||
- [ ] All `__init__` methods document their parameters
|
||||
- [ ] All public methods have docstrings with Args/Returns/Raises as needed
|
||||
- [ ] Dataclasses use "Parameters:" section for field descriptions
|
||||
- [ ] Enums document each value in "Parameters:" section
|
||||
- [ ] Writing is concise and action-oriented
|
||||
- [ ] No documentation added to private methods (starting with `_`)
|
||||
- [ ] Existing complete docstrings were left unchanged
|
||||
128
.claude/skills/pr-description/SKILL.md
Normal file
128
.claude/skills/pr-description/SKILL.md
Normal file
@@ -0,0 +1,128 @@
|
||||
---
|
||||
name: pr-description
|
||||
description: Update a GitHub PR description with a summary of changes
|
||||
---
|
||||
|
||||
Update a GitHub pull request description based on the changes in the PR.
|
||||
|
||||
## Arguments
|
||||
|
||||
```
|
||||
/pr-description <PR_NUMBER> [--fixes <ISSUE_NUMBERS>]
|
||||
```
|
||||
|
||||
- `PR_NUMBER` (required): The pull request number to update
|
||||
- `--fixes` (optional): Comma-separated issue numbers that this PR fixes (e.g., `--fixes 123,456`)
|
||||
|
||||
Examples:
|
||||
- `/pr-description 3534`
|
||||
- `/pr-description 3534 --fixes 123`
|
||||
- `/pr-description 3534 --fixes 123,456,789`
|
||||
|
||||
## Instructions
|
||||
|
||||
1. First, gather information about the PR:
|
||||
- Use GitHub plugin to get PR details (title, current description, base branch)
|
||||
- Use local git to get commits: `git log main..HEAD --oneline`
|
||||
- Use local git to get the diff: `git diff main..HEAD`
|
||||
- Parse any `--fixes` argument for issue numbers
|
||||
|
||||
2. Check the existing PR description:
|
||||
- If it already has a complete, accurate description that reflects the changes, do nothing
|
||||
- If it's missing sections, incomplete, or outdated compared to the actual changes, proceed to update
|
||||
- If it only has the template placeholder text, generate a full description
|
||||
|
||||
3. Analyze the changes:
|
||||
- Understand the purpose of each commit
|
||||
- Identify any breaking changes (API changes, removed features, behavior changes)
|
||||
- Look for new features, bug fixes, refactoring, or documentation changes
|
||||
- Collect issue numbers from:
|
||||
- The `--fixes` argument (if provided)
|
||||
- Commit messages (patterns like "Fixes #123", "Closes #456", "Resolves #789")
|
||||
|
||||
4. Generate or update the PR description with these sections:
|
||||
|
||||
## PR Description Format
|
||||
|
||||
### Summary (always include)
|
||||
|
||||
Brief bullet points describing what changed and why. Focus on the *purpose* and *impact*, not implementation details.
|
||||
|
||||
```markdown
|
||||
## Summary
|
||||
|
||||
- Added X to enable Y
|
||||
- Fixed bug where Z would happen
|
||||
- Refactored W for better maintainability
|
||||
```
|
||||
|
||||
### Breaking Changes (include only if applicable)
|
||||
|
||||
Document any changes that affect existing users or APIs.
|
||||
|
||||
```markdown
|
||||
## Breaking Changes
|
||||
|
||||
- `ClassName.method()` now requires a `param` argument
|
||||
- Removed deprecated `old_function()` - use `new_function()` instead
|
||||
```
|
||||
|
||||
### Testing (include when non-obvious)
|
||||
|
||||
How to verify the changes work. Skip for trivial changes.
|
||||
|
||||
```markdown
|
||||
## Testing
|
||||
|
||||
- Run `uv run pytest tests/test_feature.py` to verify the fix
|
||||
- Example usage: `uv run examples/new_feature.py`
|
||||
```
|
||||
|
||||
### Fixes (include if issues are provided or found in commits)
|
||||
|
||||
List issues this PR fixes. GitHub will automatically close these issues when the PR is merged.
|
||||
|
||||
```markdown
|
||||
## Fixes
|
||||
|
||||
- Fixes #123
|
||||
- Fixes #456
|
||||
```
|
||||
|
||||
Note: Use "Fixes #X" format (not "Closes" or "Resolves") for consistency. Each issue should be on its own line with "Fixes" to ensure GitHub auto-closes them.
|
||||
|
||||
## Guidelines
|
||||
|
||||
- **Be concise** - Reviewers should understand the PR in 30 seconds
|
||||
- **Focus on why** - The diff shows *what* changed, explain *why*
|
||||
- **Skip empty sections** - Only include sections that have content
|
||||
- **Use bullet points** - Easier to scan than paragraphs
|
||||
- **Don't duplicate the diff** - Avoid listing every file or line changed
|
||||
|
||||
## Example Output
|
||||
|
||||
```markdown
|
||||
## Summary
|
||||
|
||||
- Added `/docstring` skill for documenting Python modules with Google-style docstrings
|
||||
- Skill finds classes by name and handles conflicts when multiple matches exist
|
||||
- Skips already-documented code to avoid unnecessary changes
|
||||
|
||||
## Testing
|
||||
|
||||
/docstring ClassName
|
||||
|
||||
## Fixes
|
||||
|
||||
- Fixes #123
|
||||
```
|
||||
|
||||
## Checklist
|
||||
|
||||
Before updating the PR:
|
||||
|
||||
- [ ] Verified existing description needs updating (not already complete)
|
||||
- [ ] Summary accurately reflects the changes
|
||||
- [ ] Breaking changes are clearly documented (if any)
|
||||
- [ ] No unnecessary sections included
|
||||
- [ ] Description is concise and scannable
|
||||
2
.github/workflows/coverage.yaml
vendored
2
.github/workflows/coverage.yaml
vendored
@@ -33,7 +33,7 @@ jobs:
|
||||
|
||||
- name: Install dependencies
|
||||
run: |
|
||||
uv sync --group dev --extra anthropic --extra aws --extra google --extra langchain --extra websocket
|
||||
uv sync --group dev --extra anthropic --extra aws --extra google --extra langchain --extra livekit --extra websocket
|
||||
|
||||
- name: Run tests with coverage
|
||||
run: |
|
||||
|
||||
2
.github/workflows/tests.yaml
vendored
2
.github/workflows/tests.yaml
vendored
@@ -37,7 +37,7 @@ jobs:
|
||||
|
||||
- name: Install dependencies
|
||||
run: |
|
||||
uv sync --group dev --extra anthropic --extra aws --extra google --extra langchain --extra websocket
|
||||
uv sync --group dev --extra anthropic --extra aws --extra google --extra langchain --extra livekit --extra websocket
|
||||
|
||||
- name: Test with pytest
|
||||
run: |
|
||||
|
||||
11
.gitignore
vendored
11
.gitignore
vendored
@@ -4,7 +4,14 @@ __pycache__/
|
||||
*~
|
||||
venv
|
||||
.venv
|
||||
/.idea
|
||||
.idea
|
||||
.gradle
|
||||
.next
|
||||
next-env.d.ts
|
||||
local.properties
|
||||
*.log
|
||||
*.lock
|
||||
smart_turn_audio_log
|
||||
#*#
|
||||
|
||||
# Distribution / Packaging
|
||||
@@ -27,7 +34,7 @@ share/python-wheels/
|
||||
*.egg
|
||||
MANIFEST
|
||||
.DS_Store
|
||||
.env
|
||||
.env*
|
||||
fly.toml
|
||||
|
||||
# Examples
|
||||
|
||||
123
CHANGELOG.md
123
CHANGELOG.md
@@ -7,6 +7,129 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
|
||||
|
||||
<!-- towncrier release notes start -->
|
||||
|
||||
## [0.0.100] - 2026-01-20
|
||||
|
||||
### Added
|
||||
|
||||
- Added Hathora service to support Hathora-hosted TTS and STT models (only
|
||||
non-streaming)
|
||||
(PR [#3169](https://github.com/pipecat-ai/pipecat/pull/3169))
|
||||
|
||||
- Added `CambTTSService`, using Camb.ai's TTS integration with MARS models
|
||||
(mars-flash, mars-pro, mars-instruct) for high-quality text-to-speech
|
||||
synthesis.
|
||||
(PR [#3349](https://github.com/pipecat-ai/pipecat/pull/3349))
|
||||
|
||||
- Added the `additional_headers` param to `WebsocketClientParams`, allowing
|
||||
`WebsocketClientTransport` to send custom headers on connect, for cases such
|
||||
as authentication.
|
||||
(PR [#3461](https://github.com/pipecat-ai/pipecat/pull/3461))
|
||||
|
||||
- Added `UserIdleController` for detecting user idle state, integrated into
|
||||
`LLMUserAggregator` and `UserTurnProcessor` via optional `user_idle_timeout`
|
||||
parameter. Emits `on_user_turn_idle` event for application-level handling.
|
||||
Deprecated `UserIdleProcessor` in favor of the new compositional approach.
|
||||
(PR [#3482](https://github.com/pipecat-ai/pipecat/pull/3482))
|
||||
|
||||
- Added `on_user_mute_started` and `on_user_mute_stopped` event handlers to
|
||||
`LLMUserAggregator` for tracking user mute state changes.
|
||||
(PR [#3490](https://github.com/pipecat-ai/pipecat/pull/3490))
|
||||
|
||||
### Changed
|
||||
|
||||
- Enhanced interruption handling in `AsyncAITTSService` by supporting
|
||||
multi-context WebSocket sessions for more robust context management.
|
||||
(PR [#3287](https://github.com/pipecat-ai/pipecat/pull/3287))
|
||||
|
||||
- Throttle `UserSpeakingFrame` to broadcast at most every 200ms instead of on
|
||||
every audio chunk, reducing frame processing overhead during user speech.
|
||||
(PR [#3483](https://github.com/pipecat-ai/pipecat/pull/3483))
|
||||
|
||||
### Deprecated
|
||||
|
||||
- For consistency with other package names, we just deprecated
|
||||
`pipecat.turns.mute` (introduced in Pipecat 0.0.99) in favor of
|
||||
`pipecat.turns.user_mute`.
|
||||
(PR [#3479](https://github.com/pipecat-ai/pipecat/pull/3479))
|
||||
|
||||
### Fixed
|
||||
|
||||
- Corrected TTFB metric calculation in `AsyncAIHttpTTSService`.
|
||||
(PR [#3287](https://github.com/pipecat-ai/pipecat/pull/3287))
|
||||
|
||||
- Fixed an issue where the "bot-llm-text" RTVI event would not fire for
|
||||
realtime (speech-to-speech) services:
|
||||
|
||||
- `AWSNovaSonicLLMService`
|
||||
- `GeminiLiveLLMService`
|
||||
- `OpenAIRealtimeLLMService`
|
||||
- `GrokRealtimeLLMService`
|
||||
|
||||
The issue was that these services weren't pushing `LLMTextFrame`s. Now
|
||||
they do.
|
||||
(PR [#3446](https://github.com/pipecat-ai/pipecat/pull/3446))
|
||||
|
||||
- Fixed an issue where `on_user_turn_stop_timeout` could fire while a user is
|
||||
talking when using `ExternalUserTurnStrategies`.
|
||||
(PR [#3454](https://github.com/pipecat-ai/pipecat/pull/3454))
|
||||
|
||||
- Fixed an issue where user turn start strategies were not being reset after a
|
||||
user turn started, causing incorrect strategy behavior.
|
||||
(PR [#3455](https://github.com/pipecat-ai/pipecat/pull/3455))
|
||||
|
||||
- Fixed `MinWordsUserTurnStartStrategy` to not aggregate transcriptions,
|
||||
preventing incorrect turn starts when words are spoken with pauses between
|
||||
them.
|
||||
(PR [#3462](https://github.com/pipecat-ai/pipecat/pull/3462))
|
||||
|
||||
- Fixed an issue where Grok Realtime would error out when running with
|
||||
SmallWebRTC transport.
|
||||
(PR [#3480](https://github.com/pipecat-ai/pipecat/pull/3480))
|
||||
|
||||
- Fixed a `Mem0MemoryService` issue where passing `async_mode: true` was
|
||||
causing an error. See
|
||||
https://docs.mem0.ai/platform/features/async-mode-default-change.
|
||||
(PR [#3484](https://github.com/pipecat-ai/pipecat/pull/3484))
|
||||
|
||||
- Fixed `AWSNovaSonicLLMService.reset_conversation()`, which would previously
|
||||
error out. Now it successfully reconnects and "rehydrates" from the context
|
||||
object.
|
||||
(PR [#3486](https://github.com/pipecat-ai/pipecat/pull/3486))
|
||||
|
||||
- Fixed `AzureTTSService` transcript formatting issues:
|
||||
- Punctuation now appears without extra spaces (e.g., "Hello!" instead of
|
||||
"Hello !")
|
||||
- CJK languages (Chinese, Japanese, Korean) no longer have unwanted spaces
|
||||
between characters
|
||||
(PR [#3489](https://github.com/pipecat-ai/pipecat/pull/3489))
|
||||
|
||||
- Fixed an issue where `UninterruptibleFrame` frames would not be preserved in
|
||||
some cases.
|
||||
(PR [#3494](https://github.com/pipecat-ai/pipecat/pull/3494))
|
||||
|
||||
- Fixed memory leak in `LiveKitTransport` when `video_in_enabled` is `False`.
|
||||
(PR [#3499](https://github.com/pipecat-ai/pipecat/pull/3499))
|
||||
|
||||
- Fixed an issue in `AIService` where unhandled exceptions in `start()`,
|
||||
`stop()`, or `cancel()` implementations would prevent `process_frame()` to
|
||||
continue and therefore `StartFrame`, `EndFrame`, or `CancelFrame` from being
|
||||
pushed downstream, causing the pipeline to not start or stop properly.
|
||||
(PR [#3503](https://github.com/pipecat-ai/pipecat/pull/3503))
|
||||
|
||||
- Moved `NVIDIATTSService` and `NVIDIASTTService` client initialization from
|
||||
constructor to `start()` for better error handling.
|
||||
(PR [#3504](https://github.com/pipecat-ai/pipecat/pull/3504))
|
||||
|
||||
- Optimized `NVIDIATTSService` to process incoming audio frames immediately.
|
||||
(PR [#3509](https://github.com/pipecat-ai/pipecat/pull/3509))
|
||||
|
||||
- Optimized `NVIDIASTTService` by removing unnecessary queue and task.
|
||||
(PR [#3509](https://github.com/pipecat-ai/pipecat/pull/3509))
|
||||
|
||||
- Fixed a `CambTTSService` issue where client was being initialized in the
|
||||
constructor which wouldn't allow for proper Pipeline error handling.
|
||||
(PR [#3511](https://github.com/pipecat-ai/pipecat/pull/3511))
|
||||
|
||||
## [0.0.99] - 2026-01-13
|
||||
|
||||
### Added
|
||||
|
||||
@@ -81,7 +81,7 @@ Catch new features, interviews, and how-tos on our [Pipecat TV](https://www.yout
|
||||
| Serializers | [Exotel](https://docs.pipecat.ai/server/utilities/serializers/exotel), [Plivo](https://docs.pipecat.ai/server/utilities/serializers/plivo), [Twilio](https://docs.pipecat.ai/server/utilities/serializers/twilio), [Telnyx](https://docs.pipecat.ai/server/utilities/serializers/telnyx), [Vonage](https://docs.pipecat.ai/server/utilities/serializers/vonage) |
|
||||
| Video | [HeyGen](https://docs.pipecat.ai/server/services/video/heygen), [Tavus](https://docs.pipecat.ai/server/services/video/tavus), [Simli](https://docs.pipecat.ai/server/services/video/simli) |
|
||||
| Memory | [mem0](https://docs.pipecat.ai/server/services/memory/mem0) |
|
||||
| Vision & Image | [fal](https://docs.pipecat.ai/server/services/image-generation/fal), [Google Imagen](https://docs.pipecat.ai/server/services/image-generation/fal), [Moondream](https://docs.pipecat.ai/server/services/vision/moondream) |
|
||||
| Vision & Image | [fal](https://docs.pipecat.ai/server/services/image-generation/fal), [Google Imagen](https://docs.pipecat.ai/server/services/image-generation/google-imagen), [Moondream](https://docs.pipecat.ai/server/services/vision/moondream) |
|
||||
| Audio Processing | [Silero VAD](https://docs.pipecat.ai/server/utilities/audio/silero-vad-analyzer), [Krisp](https://docs.pipecat.ai/server/utilities/audio/krisp-filter), [Koala](https://docs.pipecat.ai/server/utilities/audio/koala-filter), [ai-coustics](https://docs.pipecat.ai/server/utilities/audio/aic-filter) |
|
||||
| Analytics & Metrics | [OpenTelemetry](https://docs.pipecat.ai/server/utilities/opentelemetry), [Sentry](https://docs.pipecat.ai/server/services/analytics/sentry) |
|
||||
|
||||
|
||||
@@ -1 +0,0 @@
|
||||
- Added Hathora service to support Hathora-hosted TTS and STT models (only non-streaming)
|
||||
@@ -1 +0,0 @@
|
||||
- Enhanced interruption handling in `AsyncAITTSService` by supporting multi-context WebSocket sessions for more robust context management.
|
||||
@@ -1 +0,0 @@
|
||||
- Corrected TTFB metric calculation in `AsyncAIHttpTTSService`.
|
||||
@@ -1 +0,0 @@
|
||||
- Added `CambTTSService`, using Camb.ai's TTS integration with MARS models (mars-flash, mars-pro, mars-instruct) for high-quality text-to-speech synthesis.
|
||||
@@ -1,8 +0,0 @@
|
||||
- Fixed an issue where the "bot-llm-text" RTVI event would not fire for realtime (speech-to-speech) services:
|
||||
|
||||
- `AWSNovaSonicLLMService`
|
||||
- `GeminiLiveLLMService`
|
||||
- `OpenAIRealtimeLLMService`
|
||||
- `GrokRealtimeLLMService`
|
||||
|
||||
The issue was that these services weren't pushing `LLMTextFrame`s. Now they do.
|
||||
@@ -1 +0,0 @@
|
||||
- Fixed an issue where `on_user_turn_stop_timeout` could fire while a user is talking when using `ExternalUserTurnStrategies`.
|
||||
@@ -1 +0,0 @@
|
||||
- Fixed an issue where user turn start strategies were not being reset after a user turn started, causing incorrect strategy behavior.
|
||||
@@ -1 +0,0 @@
|
||||
- Added the `additional_headers` param to `WebsocketClientParams`, allowing `WebsocketClientTransport` to send custom headers on connect, for cases such as authentication.
|
||||
@@ -1 +0,0 @@
|
||||
- Fixed `MinWordsUserTurnStartStrategy` to not aggregate transcriptions, preventing incorrect turn starts when words are spoken with pauses between them.
|
||||
@@ -1 +0,0 @@
|
||||
- For consistency with other package names, we just deprecated `pipecat.turns.mute` (introduced in Pipecat 0.0.99) in favor of `pipecat.turns.user_mute`.
|
||||
@@ -1 +0,0 @@
|
||||
- Fixed an issue where Grok Realtime would error out when running with SmallWebRTC transport.
|
||||
@@ -1 +0,0 @@
|
||||
- Added `UserIdleController` for detecting user idle state, integrated into `LLMUserAggregator` and `UserTurnProcessor` via optional `user_idle_timeout` parameter. Emits `on_user_turn_idle` event for application-level handling. Deprecated `UserIdleProcessor` in favor of the new compositional approach.
|
||||
@@ -1 +0,0 @@
|
||||
- Throttle `UserSpeakingFrame` to broadcast at most every 200ms instead of on every audio chunk, reducing frame processing overhead during user speech.
|
||||
@@ -1 +0,0 @@
|
||||
- Fixed a `Mem0MemoryService` issue where passing `async_mode: true` was causing an error. See https://docs.mem0.ai/platform/features/async-mode-default-change.
|
||||
@@ -1,3 +0,0 @@
|
||||
- Fixed `AzureTTSService` transcript formatting issues:
|
||||
- Punctuation now appears without extra spaces (e.g., "Hello!" instead of "Hello !")
|
||||
- CJK languages (Chinese, Japanese, Korean) no longer have unwanted spaces between characters
|
||||
@@ -1 +0,0 @@
|
||||
- Added `on_user_mute_started` and `on_user_mute_stopped` event handlers to `LLMUserAggregator` for tracking user mute state changes.
|
||||
1
changelog/3510.added.2.md
Normal file
1
changelog/3510.added.2.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `add_reached_upstream_filter()` and `add_reached_downstream_filter()` methods to `PipelineTask` for appending frame types.
|
||||
1
changelog/3510.added.md
Normal file
1
changelog/3510.added.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `reached_upstream_types` and `reached_downstream_types` read-only properties to `PipelineTask` for inspecting current frame filters.
|
||||
1
changelog/3510.changed.3.md
Normal file
1
changelog/3510.changed.3.md
Normal file
@@ -0,0 +1 @@
|
||||
- Changed frame filter storage from tuples to sets in `PipelineTask`.
|
||||
1
changelog/3519.added.2.md
Normal file
1
changelog/3519.added.2.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `RTVIProcessor.create_rtvi_observer()` factory method for creating RTVI observers.
|
||||
1
changelog/3519.added.3.md
Normal file
1
changelog/3519.added.3.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `FrameProcessor.broadcast_frame_instance(frame)` method to broadcast a frame instance by extracting its fields and creating new instances for each direction.
|
||||
1
changelog/3519.added.md
Normal file
1
changelog/3519.added.md
Normal file
@@ -0,0 +1 @@
|
||||
- `PipelineTask` now automatically adds `RTVIProcessor` and registers `RTVIObserver` when `enable_rtvi=True` (default), simplifying pipeline setup.
|
||||
1
changelog/3519.fixed.2.md
Normal file
1
changelog/3519.fixed.2.md
Normal file
@@ -0,0 +1 @@
|
||||
- Fixed `FrameProcessor.broadcast_frame()` to deep copy kwargs, preventing shared mutable references between the downstream and upstream frame instances.
|
||||
1
changelog/3519.fixed.md
Normal file
1
changelog/3519.fixed.md
Normal file
@@ -0,0 +1 @@
|
||||
- Transports now properly broadcast `InputTransportMessageFrame` frames both upstream and downstream instead of only pushing downstream.
|
||||
1
changelog/3520.added.md
Normal file
1
changelog/3520.added.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `video_out_codec` parameter to `TransportParams` allowing configuration of the preferred video codec (e.g., `"VP8"`, `"H264"`, `"H265"`) for video output in `DailyTransport`.
|
||||
1
changelog/3523.added.md
Normal file
1
changelog/3523.added.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added `location` parameter to Google TTS services (`GoogleHttpTTSService`, `GoogleTTSService`, `GeminiTTSService`) for regional endpoint support.
|
||||
1
changelog/3525.added.md
Normal file
1
changelog/3525.added.md
Normal file
@@ -0,0 +1 @@
|
||||
- Added new `SMART_TURN_LOG_DATA` environment variable, which causes Smart Turn input data to be saved to disk
|
||||
2
changelog/3531.changed.md
Normal file
2
changelog/3531.changed.md
Normal file
@@ -0,0 +1,2 @@
|
||||
- Changed default Inworld TTS model from `inworld-tts-1` to
|
||||
`inworld-tts-1.5-max`.
|
||||
@@ -10,7 +10,6 @@ import os
|
||||
from dotenv import load_dotenv
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
|
||||
from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnalyzerV3
|
||||
from pipecat.audio.vad.silero import SileroVADAnalyzer
|
||||
from pipecat.audio.vad.vad_analyzer import VADParams
|
||||
|
||||
@@ -45,7 +45,6 @@ from pipecat.services.google.tts import GoogleTTSService
|
||||
from pipecat.transcriptions.language import Language
|
||||
from pipecat.transports.base_transport import BaseTransport, TransportParams
|
||||
from pipecat.transports.daily.transport import DailyParams
|
||||
from pipecat.transports.websocket.fastapi import FastAPIWebsocketParams
|
||||
from pipecat.turns.user_stop import TurnAnalyzerUserTurnStopStrategy
|
||||
from pipecat.turns.user_turn_strategies import UserTurnStrategies
|
||||
|
||||
|
||||
@@ -28,7 +28,7 @@ from dotenv import load_dotenv
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.audio.filters.krisp_viva_filter import KrispVivaFilter
|
||||
from pipecat.audio.turn.krisp_viva_turn import KrispTurnParams, KrispVivaTurn
|
||||
from pipecat.audio.turn.krisp_viva_turn import KrispVivaTurn
|
||||
from pipecat.audio.vad.silero import SileroVADAnalyzer
|
||||
from pipecat.audio.vad.vad_analyzer import VADParams
|
||||
from pipecat.frames.frames import LLMRunFrame
|
||||
|
||||
@@ -23,7 +23,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.deepgram.stt import DeepgramSTTService
|
||||
@@ -93,12 +92,9 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
)
|
||||
|
||||
rtvi = RTVIProcessor()
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
transport.input(),
|
||||
rtvi,
|
||||
stt,
|
||||
user_aggregator,
|
||||
llm,
|
||||
@@ -115,7 +111,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
enable_usage_metrics=True,
|
||||
),
|
||||
observers=[
|
||||
RTVIObserver(rtvi),
|
||||
DebugLogObserver(
|
||||
frame_types={
|
||||
TTSTextFrame: (BaseOutputTransport, FrameEndpoint.SOURCE),
|
||||
|
||||
@@ -22,7 +22,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.deepgram.stt import DeepgramSTTService
|
||||
@@ -88,12 +87,9 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
)
|
||||
|
||||
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
transport.input(),
|
||||
rtvi,
|
||||
stt,
|
||||
user_aggregator,
|
||||
llm,
|
||||
@@ -110,7 +106,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
enable_usage_metrics=True,
|
||||
),
|
||||
observers=[
|
||||
RTVIObserver(rtvi),
|
||||
DebugLogObserver(
|
||||
frame_types={
|
||||
TTSTextFrame: (BaseOutputTransport, FrameEndpoint.SOURCE),
|
||||
|
||||
@@ -22,7 +22,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.deepgram.stt import DeepgramSTTService
|
||||
@@ -90,12 +89,9 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
)
|
||||
|
||||
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
transport.input(), # Transport user input
|
||||
rtvi,
|
||||
stt,
|
||||
user_aggregator, # User responses
|
||||
llm, # LLM
|
||||
@@ -114,7 +110,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
idle_timeout_secs=runner_args.pipeline_idle_timeout_secs,
|
||||
observers=[
|
||||
RTVIObserver(rtvi),
|
||||
DebugLogObserver(
|
||||
frame_types={
|
||||
TTSTextFrame: (BaseOutputTransport, FrameEndpoint.SOURCE),
|
||||
@@ -123,10 +118,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
],
|
||||
)
|
||||
|
||||
@rtvi.event_handler("on_client_ready")
|
||||
async def on_client_ready(rtvi):
|
||||
await rtvi.set_bot_ready()
|
||||
|
||||
@transport.event_handler("on_client_connected")
|
||||
async def on_client_connected(transport, client):
|
||||
logger.info(f"Client connected")
|
||||
|
||||
@@ -22,7 +22,7 @@ from pipecat.frames.frames import (
|
||||
)
|
||||
from pipecat.pipeline.pipeline import Pipeline
|
||||
from pipecat.pipeline.runner import PipelineRunner
|
||||
from pipecat.pipeline.task import PipelineParams, PipelineTask
|
||||
from pipecat.pipeline.task import PipelineTask
|
||||
from pipecat.processors.aggregators.llm_context import LLMContext
|
||||
from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
|
||||
@@ -17,7 +17,7 @@ from pipecat.adapters.schemas.tools_schema import ToolsSchema
|
||||
from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnalyzerV3
|
||||
from pipecat.audio.vad.silero import SileroVADAnalyzer
|
||||
from pipecat.audio.vad.vad_analyzer import VADParams
|
||||
from pipecat.frames.frames import LLMRunFrame, TTSSpeakFrame, UserImageRequestFrame
|
||||
from pipecat.frames.frames import LLMRunFrame, UserImageRequestFrame
|
||||
from pipecat.pipeline.pipeline import Pipeline
|
||||
from pipecat.pipeline.runner import PipelineRunner
|
||||
from pipecat.pipeline.task import PipelineParams, PipelineTask
|
||||
|
||||
@@ -22,7 +22,6 @@ from pipecat.pipeline.runner import PipelineRunner
|
||||
from pipecat.pipeline.task import PipelineParams, PipelineTask
|
||||
from pipecat.processors.aggregators.llm_context import LLMContext
|
||||
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
|
||||
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.aws.nova_sonic.llm import AWSNovaSonicLLMService
|
||||
@@ -114,6 +113,14 @@ async def load_conversation(params: FunctionCallParams):
|
||||
# "content": f"{AWSNovaSonicLLMService.AWAIT_TRIGGER_ASSISTANT_RESPONSE_INSTRUCTION}",
|
||||
# }
|
||||
# )
|
||||
# If the last message isn't from the user, add a message asking for a recap
|
||||
if messages and messages[-1].get("role") != "user":
|
||||
messages.append(
|
||||
{
|
||||
"role": "user",
|
||||
"content": "Can you catch me up on what we were talking about?",
|
||||
}
|
||||
)
|
||||
params.context.set_messages(messages)
|
||||
await params.llm.reset_conversation()
|
||||
# await params.llm.trigger_assistant_response()
|
||||
|
||||
@@ -59,7 +59,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.deepgram.stt import DeepgramSTTService
|
||||
@@ -255,12 +254,10 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
),
|
||||
)
|
||||
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
transport.input(),
|
||||
rtvi,
|
||||
stt,
|
||||
user_aggregator,
|
||||
memory,
|
||||
@@ -278,12 +275,10 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
enable_usage_metrics=True,
|
||||
),
|
||||
idle_timeout_secs=runner_args.pipeline_idle_timeout_secs,
|
||||
observers=[RTVIObserver(rtvi)],
|
||||
)
|
||||
|
||||
@rtvi.event_handler("on_client_ready")
|
||||
@task.rtvi.event_handler("on_client_ready")
|
||||
async def on_client_ready(rtvi):
|
||||
await rtvi.set_bot_ready()
|
||||
# Get personalized greeting based on user memories. Can pass agent_id and run_id as per requirement of the application to manage short term memory or agent specific memory.
|
||||
greeting = await get_initial_greeting(
|
||||
memory_client=memory.memory_client, user_id=USER_ID, agent_id=None, run_id=None
|
||||
|
||||
@@ -22,7 +22,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMContextAggregatorPair,
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.cartesia.tts import CartesiaTTSService
|
||||
@@ -87,8 +86,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
),
|
||||
)
|
||||
|
||||
rtvi = RTVIProcessor()
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
transport.input(), # Transport user input
|
||||
@@ -108,13 +105,11 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
|
||||
enable_metrics=True,
|
||||
enable_usage_metrics=True,
|
||||
),
|
||||
observers=[RTVIObserver(rtvi)],
|
||||
idle_timeout_secs=runner_args.pipeline_idle_timeout_secs,
|
||||
)
|
||||
|
||||
@rtvi.event_handler("on_client_ready")
|
||||
@task.rtvi.event_handler("on_client_ready")
|
||||
async def on_client_ready(rtvi):
|
||||
await rtvi.set_bot_ready()
|
||||
# Kick off the conversation
|
||||
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
|
||||
await task.queue_frames([LLMRunFrame()])
|
||||
|
||||
@@ -9,7 +9,6 @@ import asyncio
|
||||
import io
|
||||
import json
|
||||
import os
|
||||
import re
|
||||
import shutil
|
||||
|
||||
import aiohttp
|
||||
|
||||
@@ -1,5 +1,5 @@
|
||||
#
|
||||
# Copyright (c) 2025, Daily
|
||||
# Copyright (c) 2024-2026, Daily
|
||||
#
|
||||
# SPDX-License-Identifier: BSD 2-Clause License
|
||||
#
|
||||
@@ -22,7 +22,6 @@ from pipecat.processors.aggregators.llm_response_universal import (
|
||||
LLMUserAggregatorParams,
|
||||
)
|
||||
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIProcessor
|
||||
from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.google.gemini_live.llm import GeminiLiveLLMService
|
||||
@@ -125,14 +124,10 @@ async def run_bot(pipecat_transport):
|
||||
),
|
||||
)
|
||||
|
||||
# RTVI events for Pipecat client UI
|
||||
rtvi = RTVIProcessor()
|
||||
|
||||
pipeline = Pipeline(
|
||||
[
|
||||
pipecat_transport.input(),
|
||||
user_aggregator,
|
||||
rtvi,
|
||||
llm, # LLM
|
||||
EdgeDetectionProcessor(
|
||||
pipecat_transport._params.video_out_width,
|
||||
@@ -149,13 +144,11 @@ async def run_bot(pipecat_transport):
|
||||
enable_metrics=True,
|
||||
enable_usage_metrics=True,
|
||||
),
|
||||
observers=[RTVIObserver(rtvi)],
|
||||
)
|
||||
|
||||
@rtvi.event_handler("on_client_ready")
|
||||
@task.rtvi.event_handler("on_client_ready")
|
||||
async def on_client_ready(rtvi):
|
||||
logger.info("Pipecat client ready.")
|
||||
await rtvi.set_bot_ready()
|
||||
# Kick off the conversation.
|
||||
await task.queue_frames([LLMRunFrame()])
|
||||
|
||||
|
||||
@@ -13,7 +13,7 @@ from pipecat.adapters.schemas.tools_schema import ToolsSchema
|
||||
from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnalyzerV3
|
||||
from pipecat.audio.vad.silero import SileroVADAnalyzer
|
||||
from pipecat.audio.vad.vad_analyzer import VADParams
|
||||
from pipecat.frames.frames import LLMRunFrame, ThoughtTranscriptionMessage, TranscriptionMessage
|
||||
from pipecat.frames.frames import LLMRunFrame
|
||||
from pipecat.pipeline.pipeline import Pipeline
|
||||
from pipecat.pipeline.runner import PipelineRunner
|
||||
from pipecat.pipeline.task import PipelineParams, PipelineTask
|
||||
|
||||
@@ -53,8 +53,6 @@ from pipecat.runner.types import RunnerArguments
|
||||
from pipecat.runner.utils import create_transport
|
||||
from pipecat.services.grok.realtime.events import (
|
||||
SessionProperties,
|
||||
WebSearchTool,
|
||||
XSearchTool,
|
||||
)
|
||||
from pipecat.services.grok.realtime.llm import GrokRealtimeLLMService
|
||||
from pipecat.services.llm_service import FunctionCallParams
|
||||
|
||||
@@ -1,11 +1,11 @@
|
||||
agent_name = "quickstart"
|
||||
image = "your_username/quickstart:0.1"
|
||||
secret_set = "quickstart-secrets"
|
||||
agent_name = "quickstart-test"
|
||||
image = "markatdaily/quickstart-test:latest"
|
||||
secret_set = "quickstart-test-secrets"
|
||||
agent_profile = "agent-1x"
|
||||
|
||||
# RECOMMENDED: Set an image pull secret:
|
||||
# https://docs.pipecat.ai/deployment/pipecat-cloud/fundamentals/secrets#image-pull-secrets
|
||||
# image_credentials = "your_image_pull_secret"
|
||||
image_credentials = "dockerhub-access"
|
||||
|
||||
[scaling]
|
||||
min_agents = 1
|
||||
|
||||
@@ -293,12 +293,13 @@ async def run_eval_pipeline(
|
||||
"You should only call the eval function if:\n"
|
||||
"- The user explicitly attempts to answer the question, AND\n"
|
||||
f"- Their answer can be cleanly evaluated using: {eval_config.eval}\n"
|
||||
"Ignore greetings, comments, non-answers, or requests for clarification."
|
||||
"Ignore greetings, comments, non-answers, or requests for clarification.\n"
|
||||
"Numerical word answers are allowed (e.g., 'five' is the same as '5').\n"
|
||||
)
|
||||
if eval_config.eval_speaks_first:
|
||||
system_prompt = f"You are an evaluation agent, be extremly brief. Numerical word answers are allowed. You will start the conversation by saying: '{example_prompt}'. {common_system_prompt}"
|
||||
system_prompt = f"You are an evaluation agent, be extremly brief. You will start the conversation by saying: '{example_prompt}'. {common_system_prompt}"
|
||||
else:
|
||||
system_prompt = f"You are an evaluation agent, be extremly brief. Numerical word answers are allowed. First, ask one question: {example_prompt}. {common_system_prompt}"
|
||||
system_prompt = f"You are an evaluation agent, be extremly brief. First, ask one question: {example_prompt}. {common_system_prompt}"
|
||||
|
||||
messages = [
|
||||
{
|
||||
|
||||
@@ -137,6 +137,7 @@ TESTS_07 = [
|
||||
# ("07zd-interruptible-aicoustics.py", EVAL_SIMPLE_MATH),
|
||||
("07ze-interruptible-hume.py", EVAL_SIMPLE_MATH),
|
||||
("07zf-interruptible-gradium.py", EVAL_SIMPLE_MATH),
|
||||
("07zg-interruptible-camb.py", EVAL_SIMPLE_MATH),
|
||||
("07zh-interruptible-hathora.py", EVAL_SIMPLE_MATH),
|
||||
# Needs a local XTTS docker instance running.
|
||||
# ("07i-interruptible-xtts.py", EVAL_SIMPLE_MATH),
|
||||
|
||||
@@ -22,7 +22,7 @@ from pathlib import Path
|
||||
|
||||
try:
|
||||
import numpy as np
|
||||
import soundfile as sf
|
||||
import soundfile as sf # noqa: F401
|
||||
from audio_file_utils import calculate_audio_stats, read_audio_file, write_audio_file
|
||||
except ImportError as e:
|
||||
print(f"Error: Missing required dependencies: {e}")
|
||||
|
||||
@@ -23,7 +23,7 @@ from pathlib import Path
|
||||
|
||||
try:
|
||||
import numpy as np
|
||||
import soundfile as sf
|
||||
import soundfile as sf # noqa: F401
|
||||
from audio_file_utils import read_audio_file
|
||||
except ImportError as e:
|
||||
print(f"Error: Missing required dependencies: {e}")
|
||||
|
||||
@@ -10,7 +10,7 @@ import base64
|
||||
import copy
|
||||
import json
|
||||
from dataclasses import dataclass
|
||||
from typing import Any, Dict, List, Literal, Optional, TypedDict
|
||||
from typing import Any, Dict, List, Optional, TypedDict
|
||||
|
||||
from loguru import logger
|
||||
|
||||
|
||||
@@ -9,7 +9,7 @@
|
||||
import base64
|
||||
import json
|
||||
from dataclasses import dataclass, field
|
||||
from typing import Any, Dict, List, Optional, Tuple, TypedDict
|
||||
from typing import Any, Dict, List, Optional, TypedDict
|
||||
|
||||
from loguru import logger
|
||||
from openai import NotGiven
|
||||
|
||||
@@ -7,10 +7,8 @@
|
||||
"""OpenAI LLM adapter for Pipecat."""
|
||||
|
||||
import copy
|
||||
import json
|
||||
from typing import Any, Dict, List, TypedDict
|
||||
|
||||
from openai._types import NOT_GIVEN as OPEN_AI_NOT_GIVEN
|
||||
from openai._types import NotGiven as OpenAINotGiven
|
||||
from openai.types.chat import (
|
||||
ChatCompletionMessageParam,
|
||||
|
||||
@@ -9,7 +9,6 @@
|
||||
This module provides an audio filter implementation using Krisp VIVA SDK.
|
||||
"""
|
||||
|
||||
import asyncio
|
||||
import os
|
||||
|
||||
import numpy as np
|
||||
|
||||
@@ -16,6 +16,7 @@ import numpy as np
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.audio.turn.smart_turn.base_smart_turn import BaseSmartTurn
|
||||
from pipecat.utils.env import env_truthy
|
||||
|
||||
try:
|
||||
import onnxruntime as ort
|
||||
@@ -48,6 +49,8 @@ class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
|
||||
"""
|
||||
super().__init__(**kwargs)
|
||||
|
||||
self._log_data = env_truthy("SMART_TURN_LOG_DATA", default=False)
|
||||
|
||||
if not smart_turn_model_path:
|
||||
# Load bundled model
|
||||
model_name = "smart-turn-v3.2-cpu.onnx"
|
||||
@@ -81,6 +84,49 @@ class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
|
||||
|
||||
logger.debug("Loaded Local Smart Turn v3.x")
|
||||
|
||||
def _write_audio_to_wav(
|
||||
self, audio_array: np.ndarray, sample_rate: int = 16000, suffix: str = ""
|
||||
) -> None:
|
||||
"""Write audio data to a WAV file in a background thread.
|
||||
|
||||
Args:
|
||||
audio_array: The audio data as a numpy array (float32, normalized to [-1, 1]).
|
||||
sample_rate: The sample rate of the audio data.
|
||||
suffix: Optional suffix to append to the filename (e.g., "_raw", "_padded").
|
||||
"""
|
||||
import os
|
||||
import threading
|
||||
import wave
|
||||
from datetime import datetime
|
||||
|
||||
# Generate filename with current timestamp (millisecond precision)
|
||||
timestamp = datetime.now().strftime("%Y-%m-%d__%H:%M:%S.%f")[:-3]
|
||||
log_dir = "./smart_turn_audio_log"
|
||||
os.makedirs(log_dir, exist_ok=True)
|
||||
filename = os.path.join(log_dir, f"{timestamp}{suffix}.wav")
|
||||
|
||||
# Make a copy of the audio data to avoid issues with the array being modified
|
||||
audio_copy = audio_array.copy()
|
||||
|
||||
def write_wav():
|
||||
try:
|
||||
# Convert float32 audio to int16 for WAV file
|
||||
audio_int16 = (audio_copy * 32767).astype(np.int16)
|
||||
|
||||
with wave.open(filename, "wb") as wav_file:
|
||||
wav_file.setnchannels(1) # Mono
|
||||
wav_file.setsampwidth(2) # 2 bytes for int16
|
||||
wav_file.setframerate(sample_rate)
|
||||
wav_file.writeframes(audio_int16.tobytes())
|
||||
|
||||
logger.debug(f"Wrote audio to {filename}")
|
||||
except Exception as e:
|
||||
logger.error(f"Failed to write audio to {filename}: {e}")
|
||||
|
||||
# Start background thread to write the WAV file
|
||||
thread = threading.Thread(target=write_wav, daemon=True)
|
||||
thread.start()
|
||||
|
||||
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
|
||||
"""Predict end-of-turn using local ONNX model."""
|
||||
|
||||
@@ -95,6 +141,8 @@ class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
|
||||
return np.pad(audio_array, (padding, 0), mode="constant", constant_values=0)
|
||||
return audio_array
|
||||
|
||||
audio_for_logging = audio_array
|
||||
|
||||
# Truncate to 8 seconds (keeping the end) or pad to 8 seconds
|
||||
audio_array = truncate_audio_to_last_n_seconds(audio_array, n_seconds=8)
|
||||
|
||||
@@ -122,6 +170,10 @@ class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
|
||||
# Make prediction (1 for Complete, 0 for Incomplete)
|
||||
prediction = 1 if probability > 0.5 else 0
|
||||
|
||||
if self._log_data:
|
||||
suffix = "_complete" if prediction == 1 else "_incomplete"
|
||||
self._write_audio_to_wav(audio_for_logging, sample_rate=16000, suffix=suffix)
|
||||
|
||||
return {
|
||||
"prediction": prediction,
|
||||
"probability": probability,
|
||||
|
||||
@@ -15,7 +15,7 @@ import asyncio
|
||||
import importlib.util
|
||||
import os
|
||||
from pathlib import Path
|
||||
from typing import Any, AsyncIterable, Dict, Iterable, List, Optional, Tuple, Type
|
||||
from typing import Any, AsyncIterable, Dict, Iterable, List, Optional, Set, Tuple, Type
|
||||
|
||||
from loguru import logger
|
||||
from pydantic import BaseModel, ConfigDict, Field
|
||||
@@ -49,6 +49,7 @@ from pipecat.pipeline.pipeline import Pipeline, PipelineSink, PipelineSource
|
||||
from pipecat.pipeline.task_observer import TaskObserver
|
||||
from pipecat.processors.aggregators.llm_response import LLMUserContextAggregator
|
||||
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor, FrameProcessorSetup
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserverParams, RTVIProcessor
|
||||
from pipecat.utils.asyncio.task_manager import BaseTaskManager, TaskManager, TaskManagerParams
|
||||
from pipecat.utils.tracing.setup import is_tracing_available
|
||||
from pipecat.utils.tracing.turn_trace_observer import TurnTraceObserver
|
||||
@@ -225,9 +226,12 @@ class PipelineTask(BasePipelineTask):
|
||||
conversation_id: Optional[str] = None,
|
||||
enable_tracing: bool = False,
|
||||
enable_turn_tracking: bool = True,
|
||||
enable_rtvi: bool = True,
|
||||
idle_timeout_frames: Tuple[Type[Frame], ...] = (BotSpeakingFrame, UserSpeakingFrame),
|
||||
idle_timeout_secs: Optional[float] = IDLE_TIMEOUT_SECS,
|
||||
observers: Optional[List[BaseObserver]] = None,
|
||||
rtvi_processor: Optional[RTVIProcessor] = None,
|
||||
rtvi_observer_params: Optional[RTVIObserverParams] = None,
|
||||
task_manager: Optional[BaseTaskManager] = None,
|
||||
):
|
||||
"""Initialize the PipelineTask.
|
||||
@@ -244,6 +248,7 @@ class PipelineTask(BasePipelineTask):
|
||||
check_dangling_tasks: Whether to check for processors' tasks finishing properly.
|
||||
clock: Clock implementation for timing operations.
|
||||
conversation_id: Optional custom ID for the conversation.
|
||||
enable_rtvi: Whether to automatically add RTVI support to the pipeline.
|
||||
enable_tracing: Whether to enable tracing.
|
||||
enable_turn_tracking: Whether to enable turn tracking.
|
||||
idle_timeout_frames: A tuple with the frames that should trigger an idle
|
||||
@@ -252,6 +257,8 @@ class PipelineTask(BasePipelineTask):
|
||||
None. If a pipeline is idle the pipeline task will be cancelled
|
||||
automatically.
|
||||
observers: List of observers for monitoring pipeline execution.
|
||||
rtvi_observer_params: The RTVI observer parameter to use if RTVI is enabled.
|
||||
rtvi_processor: The RTVI processor to add if RTVI is enabled.
|
||||
task_manager: Optional task manager for handling asyncio tasks.
|
||||
"""
|
||||
super().__init__()
|
||||
@@ -306,6 +313,16 @@ class PipelineTask(BasePipelineTask):
|
||||
self._heartbeat_push_task: Optional[asyncio.Task] = None
|
||||
self._heartbeat_monitor_task: Optional[asyncio.Task] = None
|
||||
|
||||
# RTVI support
|
||||
self._rtvi = None
|
||||
if enable_rtvi:
|
||||
self._rtvi = rtvi_processor or RTVIProcessor()
|
||||
observers.append(self._rtvi.create_rtvi_observer(params=rtvi_observer_params))
|
||||
|
||||
@self.rtvi.event_handler("on_client_ready")
|
||||
async def on_client_ready(rtvi: RTVIProcessor):
|
||||
await rtvi.set_bot_ready()
|
||||
|
||||
# This is the idle event. When selected frames are pushed from any
|
||||
# processor we consider the pipeline is not idle. We use an observer
|
||||
# which will be listening any part of the pipeline.
|
||||
@@ -335,7 +352,8 @@ class PipelineTask(BasePipelineTask):
|
||||
# allows us to receive and react to downstream frames.
|
||||
source = PipelineSource(self._source_push_frame, name=f"{self}::Source")
|
||||
sink = PipelineSink(self._sink_push_frame, name=f"{self}::Sink")
|
||||
self._pipeline = Pipeline([pipeline], source=source, sink=sink)
|
||||
processors = [self._rtvi, pipeline] if self._rtvi else [pipeline]
|
||||
self._pipeline = Pipeline(processors, source=source, sink=sink)
|
||||
|
||||
# The task observer acts as a proxy to the provided observers. This way,
|
||||
# we only need to pass a single observer (using the StartFrame) which
|
||||
@@ -348,8 +366,8 @@ class PipelineTask(BasePipelineTask):
|
||||
# in. This is mainly for efficiency reason because each event handler
|
||||
# creates a task and most likely you only care about one or two frame
|
||||
# types.
|
||||
self._reached_upstream_types: Tuple[Type[Frame], ...] = ()
|
||||
self._reached_downstream_types: Tuple[Type[Frame], ...] = ()
|
||||
self._reached_upstream_types: Set[Type[Frame]] = set()
|
||||
self._reached_downstream_types: Set[Type[Frame]] = set()
|
||||
self._register_event_handler("on_frame_reached_upstream")
|
||||
self._register_event_handler("on_frame_reached_downstream")
|
||||
self._register_event_handler("on_idle_timeout")
|
||||
@@ -398,6 +416,35 @@ class PipelineTask(BasePipelineTask):
|
||||
"""
|
||||
return self._turn_trace_observer
|
||||
|
||||
@property
|
||||
def rtvi(self) -> RTVIProcessor:
|
||||
"""Get the RTVI processor if RTVI is enabled.
|
||||
|
||||
Returns:
|
||||
The RTVI processor added to the pipeline when RTVI is enabled.
|
||||
"""
|
||||
if not self._rtvi:
|
||||
raise Exception(f"{self} RTVI is not enabled.")
|
||||
return self._rtvi
|
||||
|
||||
@property
|
||||
def reached_upstream_types(self) -> Tuple[Type[Frame], ...]:
|
||||
"""Get the currently configured upstream frame type filters.
|
||||
|
||||
Returns:
|
||||
Tuple of frame types that trigger the on_frame_reached_upstream event.
|
||||
"""
|
||||
return tuple(self._reached_upstream_types)
|
||||
|
||||
@property
|
||||
def reached_downstream_types(self) -> Tuple[Type[Frame], ...]:
|
||||
"""Get the currently configured downstream frame type filters.
|
||||
|
||||
Returns:
|
||||
Tuple of frame types that trigger the on_frame_reached_downstream event.
|
||||
"""
|
||||
return tuple(self._reached_downstream_types)
|
||||
|
||||
def event_handler(self, event_name: str):
|
||||
"""Decorator for registering event handlers.
|
||||
|
||||
@@ -441,7 +488,7 @@ class PipelineTask(BasePipelineTask):
|
||||
Args:
|
||||
types: Tuple of frame types to monitor for upstream events.
|
||||
"""
|
||||
self._reached_upstream_types = types
|
||||
self._reached_upstream_types = set(types)
|
||||
|
||||
def set_reached_downstream_filter(self, types: Tuple[Type[Frame], ...]):
|
||||
"""Set which frame types trigger the on_frame_reached_downstream event.
|
||||
@@ -449,7 +496,23 @@ class PipelineTask(BasePipelineTask):
|
||||
Args:
|
||||
types: Tuple of frame types to monitor for downstream events.
|
||||
"""
|
||||
self._reached_downstream_types = types
|
||||
self._reached_downstream_types = set(types)
|
||||
|
||||
def add_reached_upstream_filter(self, types: Tuple[Type[Frame], ...]):
|
||||
"""Add frame types to trigger the on_frame_reached_upstream event.
|
||||
|
||||
Args:
|
||||
types: Tuple of frame types to add to upstream monitoring.
|
||||
"""
|
||||
self._reached_upstream_types.update(types)
|
||||
|
||||
def add_reached_downstream_filter(self, types: Tuple[Type[Frame], ...]):
|
||||
"""Add frame types to trigger the on_frame_reached_downstream event.
|
||||
|
||||
Args:
|
||||
types: Tuple of frame types to add to downstream monitoring.
|
||||
"""
|
||||
self._reached_downstream_types.update(types)
|
||||
|
||||
def has_finished(self) -> bool:
|
||||
"""Check if the pipeline task has finished execution.
|
||||
@@ -749,7 +812,7 @@ class PipelineTask(BasePipelineTask):
|
||||
pipeline to be stopped (e.g. EndTaskFrame) in which case we would send
|
||||
an EndFrame down the pipeline.
|
||||
"""
|
||||
if isinstance(frame, self._reached_upstream_types):
|
||||
if isinstance(frame, tuple(self._reached_upstream_types)):
|
||||
await self._call_event_handler("on_frame_reached_upstream", frame)
|
||||
|
||||
if isinstance(frame, EndTaskFrame):
|
||||
@@ -788,7 +851,7 @@ class PipelineTask(BasePipelineTask):
|
||||
processors have handled the EndFrame and therefore we can exit the task
|
||||
cleanly.
|
||||
"""
|
||||
if isinstance(frame, self._reached_downstream_types):
|
||||
if isinstance(frame, tuple(self._reached_downstream_types)):
|
||||
await self._call_event_handler("on_frame_reached_downstream", frame)
|
||||
|
||||
if isinstance(frame, StartFrame):
|
||||
|
||||
@@ -34,7 +34,6 @@ from PIL import Image
|
||||
from pipecat.adapters.base_llm_adapter import BaseLLMAdapter
|
||||
from pipecat.adapters.schemas.tools_schema import ToolsSchema
|
||||
from pipecat.frames.frames import AudioRawFrame, Frame
|
||||
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
|
||||
|
||||
# JSON custom encoder to handle bytes arrays so that we can log contexts
|
||||
# with images to the console.
|
||||
|
||||
@@ -18,7 +18,7 @@ from typing import List
|
||||
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.frames.frames import ErrorFrame, Frame, TranscriptionFrame
|
||||
from pipecat.frames.frames import Frame, TranscriptionFrame
|
||||
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
|
||||
|
||||
|
||||
|
||||
@@ -12,7 +12,9 @@ management, and frame flow control mechanisms.
|
||||
"""
|
||||
|
||||
import asyncio
|
||||
import dataclasses
|
||||
import traceback
|
||||
from copy import deepcopy
|
||||
from dataclasses import dataclass
|
||||
from enum import Enum
|
||||
from typing import (
|
||||
@@ -779,8 +781,40 @@ class FrameProcessor(BaseObject):
|
||||
frame_cls: The class of the frame to be broadcasted.
|
||||
**kwargs: Keyword arguments to be passed to the frame's constructor.
|
||||
"""
|
||||
await self.push_frame(frame_cls(**kwargs))
|
||||
await self.push_frame(frame_cls(**kwargs), FrameDirection.UPSTREAM)
|
||||
await self.push_frame(frame_cls(**deepcopy(kwargs)))
|
||||
await self.push_frame(frame_cls(**deepcopy(kwargs)), FrameDirection.UPSTREAM)
|
||||
|
||||
async def broadcast_frame_instance(self, frame: Frame):
|
||||
"""Broadcasts a frame instance upstream and downstream.
|
||||
|
||||
This method creates two new frame instances copying all fields from the
|
||||
original frame except `id` and `name`, which get fresh values.
|
||||
|
||||
Args:
|
||||
frame: The frame instance to broadcast.
|
||||
|
||||
Note:
|
||||
Prefer using `broadcast_frame()` when possible, as it is more
|
||||
efficient. This method should only be used when you are not the
|
||||
creator of the frame and need to broadcast an existing instance.
|
||||
"""
|
||||
frame_cls = type(frame)
|
||||
init_fields = {f.name: getattr(frame, f.name) for f in dataclasses.fields(frame) if f.init}
|
||||
extra_fields = {
|
||||
f.name: getattr(frame, f.name)
|
||||
for f in dataclasses.fields(frame)
|
||||
if not f.init and f.name not in ("id", "name")
|
||||
}
|
||||
|
||||
new_frame = frame_cls(**deepcopy(init_fields))
|
||||
for k, v in deepcopy(extra_fields).items():
|
||||
setattr(new_frame, k, v)
|
||||
await self.push_frame(new_frame)
|
||||
|
||||
new_frame = frame_cls(**deepcopy(init_fields))
|
||||
for k, v in deepcopy(extra_fields).items():
|
||||
setattr(new_frame, k, v)
|
||||
await self.push_frame(new_frame, FrameDirection.UPSTREAM)
|
||||
|
||||
async def __start(self, frame: StartFrame):
|
||||
"""Handle the start frame to initialize processor state.
|
||||
@@ -950,7 +984,8 @@ class FrameProcessor(BaseObject):
|
||||
# Process current queue and keep UninterruptibleFrame frames.
|
||||
while not self.__process_queue.empty():
|
||||
item = self.__process_queue.get_nowait()
|
||||
if isinstance(item, UninterruptibleFrame):
|
||||
frame = item[0]
|
||||
if isinstance(frame, UninterruptibleFrame):
|
||||
new_queue.put_nowait(item)
|
||||
self.__process_queue.task_done()
|
||||
|
||||
|
||||
@@ -1100,13 +1100,11 @@ class RTVIObserver(BaseObserver):
|
||||
|
||||
if (
|
||||
isinstance(frame, (UserStartedSpeakingFrame, UserStoppedSpeakingFrame))
|
||||
and (direction == FrameDirection.DOWNSTREAM)
|
||||
and self._params.user_speaking_enabled
|
||||
):
|
||||
await self._handle_interruptions(frame)
|
||||
elif (
|
||||
isinstance(frame, (BotStartedSpeakingFrame, BotStoppedSpeakingFrame))
|
||||
and (direction == FrameDirection.UPSTREAM)
|
||||
and self._params.bot_speaking_enabled
|
||||
):
|
||||
await self._handle_bot_speaking(frame)
|
||||
@@ -1413,6 +1411,18 @@ class RTVIProcessor(FrameProcessor):
|
||||
|
||||
self._registered_services[service.name] = service
|
||||
|
||||
def create_rtvi_observer(self, *, params: Optional[RTVIObserverParams] = None, **kwargs):
|
||||
"""Creates a new RTVI Observer.
|
||||
|
||||
Args:
|
||||
params: Settings to enable/disable specific messages.
|
||||
**kwargs: Additional arguments passed to the observer.
|
||||
|
||||
Returns:
|
||||
A new RTVI observer.
|
||||
"""
|
||||
return RTVIObserver(self, params=params, **kwargs)
|
||||
|
||||
async def set_client_ready(self):
|
||||
"""Mark the client as ready and trigger the ready event."""
|
||||
self._client_ready = True
|
||||
|
||||
@@ -263,7 +263,7 @@ def _setup_webrtc_routes(
|
||||
"""Handle WebRTC offer requests via SmallWebRTCRequestHandler."""
|
||||
|
||||
# Prepare runner arguments with the callback to run your bot
|
||||
async def webrtc_connection_callback(connection):
|
||||
async def webrtc_connection_callback(connection: SmallWebRTCConnection):
|
||||
bot_module = _get_bot_module()
|
||||
|
||||
runner_args = SmallWebRTCRunnerArguments(
|
||||
@@ -406,13 +406,7 @@ def _setup_whatsapp_routes(app: FastAPI):
|
||||
return
|
||||
|
||||
try:
|
||||
from pipecat_ai_small_webrtc_prebuilt.frontend import SmallWebRTCPrebuiltUI
|
||||
|
||||
from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
|
||||
from pipecat.transports.smallwebrtc.request_handler import (
|
||||
SmallWebRTCRequest,
|
||||
SmallWebRTCRequestHandler,
|
||||
)
|
||||
from pipecat.transports.whatsapp.api import WhatsAppWebhookRequest
|
||||
from pipecat.transports.whatsapp.client import WhatsAppClient
|
||||
except ImportError as e:
|
||||
|
||||
@@ -126,7 +126,7 @@ class ProtobufFrameSerializer(FrameSerializer):
|
||||
if "pts" in args_dict:
|
||||
del args_dict["pts"]
|
||||
|
||||
# Special handling for MessageFrame -> OutputTransportMessageUrgentFrame
|
||||
# Special handling for MessageFrame -> InputTransportMessageFrame
|
||||
if class_name == MessageFrame:
|
||||
try:
|
||||
msg = json.loads(args_dict["data"])
|
||||
|
||||
@@ -148,11 +148,11 @@ class AIService(FrameProcessor):
|
||||
await super().process_frame(frame, direction)
|
||||
|
||||
if isinstance(frame, StartFrame):
|
||||
await self.start(frame)
|
||||
elif isinstance(frame, CancelFrame):
|
||||
await self.cancel(frame)
|
||||
await self._start(frame)
|
||||
elif isinstance(frame, EndFrame):
|
||||
await self.stop(frame)
|
||||
await self._stop(frame)
|
||||
elif isinstance(frame, CancelFrame):
|
||||
await self._cancel(frame)
|
||||
|
||||
async def process_generator(self, generator: AsyncGenerator[Frame | None, None]):
|
||||
"""Process frames from an async generator.
|
||||
@@ -169,3 +169,21 @@ class AIService(FrameProcessor):
|
||||
await self.push_error_frame(f)
|
||||
else:
|
||||
await self.push_frame(f)
|
||||
|
||||
async def _start(self, frame: StartFrame):
|
||||
try:
|
||||
await self.start(frame)
|
||||
except Exception as e:
|
||||
logger.error(f"{self}: exception processing {frame}: {e}")
|
||||
|
||||
async def _stop(self, frame: EndFrame):
|
||||
try:
|
||||
await self.stop(frame)
|
||||
except Exception as e:
|
||||
logger.error(f"{self}: exception processing {frame}: {e}")
|
||||
|
||||
async def _cancel(self, frame: CancelFrame):
|
||||
try:
|
||||
await self.cancel(frame)
|
||||
except Exception as e:
|
||||
logger.error(f"{self}: exception processing {frame}: {e}")
|
||||
|
||||
@@ -296,6 +296,7 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
self._user_text_buffer = ""
|
||||
self._assistant_text_buffer = ""
|
||||
self._completed_tool_calls = set()
|
||||
self._audio_input_started = False
|
||||
|
||||
file_path = files("pipecat.services.aws.nova_sonic").joinpath("ready.wav")
|
||||
with wave.open(file_path.open("rb"), "rb") as wav_file:
|
||||
@@ -532,14 +533,30 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
if system_instruction:
|
||||
await self._send_text_event(text=system_instruction, role=Role.SYSTEM)
|
||||
|
||||
# Send conversation history
|
||||
for message in llm_connection_params["messages"]:
|
||||
# Send conversation history (except for the last message if it's from the
|
||||
# user, which we'll send as interactive after starting audio input)
|
||||
messages = llm_connection_params["messages"]
|
||||
last_user_message = None
|
||||
for i, message in enumerate(messages):
|
||||
# logger.debug(f"Seeding conversation history with message: {message}")
|
||||
await self._send_text_event(text=message.text, role=message.role)
|
||||
is_last_message = i == len(messages) - 1
|
||||
if is_last_message and message.role == Role.USER:
|
||||
# Save for sending after audio input starts
|
||||
last_user_message = message
|
||||
else:
|
||||
await self._send_text_event(text=message.text, role=message.role)
|
||||
|
||||
# Start audio input
|
||||
await self._send_audio_input_start_event()
|
||||
|
||||
# Now send the last user message as interactive to trigger bot response
|
||||
if last_user_message:
|
||||
# logger.debug(
|
||||
# f"Sending last user message as interactive to trigger bot response: {last_user_message}")
|
||||
await self._send_text_event(
|
||||
text=last_user_message.text, role=last_user_message.role, interactive=True
|
||||
)
|
||||
|
||||
# Start receiving events
|
||||
self._receive_task = self.create_task(self._receive_task_handler())
|
||||
|
||||
@@ -602,6 +619,7 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
self._user_text_buffer = ""
|
||||
self._assistant_text_buffer = ""
|
||||
self._completed_tool_calls = set()
|
||||
self._audio_input_started = False
|
||||
|
||||
logger.info("Finished disconnecting")
|
||||
except Exception as e:
|
||||
@@ -727,8 +745,18 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
}}
|
||||
'''
|
||||
await self._send_client_event(audio_content_start)
|
||||
self._audio_input_started = True
|
||||
|
||||
async def _send_text_event(self, text: str, role: Role):
|
||||
async def _send_text_event(self, text: str, role: Role, interactive: bool = False):
|
||||
"""Send a text event to the LLM.
|
||||
|
||||
Args:
|
||||
text: The text content to send.
|
||||
role: The role associated with the text (e.g., USER, ASSISTANT, SYSTEM).
|
||||
interactive: Whether the content is interactive. Defaults to False.
|
||||
False: conversation history or system instruction, sent prior to interactive audio
|
||||
True: text input sent during (or at the start of) interactive audio
|
||||
"""
|
||||
if not self._stream or not self._prompt_name or not text:
|
||||
return
|
||||
|
||||
@@ -741,7 +769,7 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
"promptName": "{self._prompt_name}",
|
||||
"contentName": "{content_name}",
|
||||
"type": "TEXT",
|
||||
"interactive": true,
|
||||
"interactive": {json.dumps(interactive)},
|
||||
"role": "{role.value}",
|
||||
"textInputConfiguration": {{
|
||||
"mediaType": "text/plain"
|
||||
@@ -779,7 +807,7 @@ class AWSNovaSonicLLMService(LLMService):
|
||||
await self._send_client_event(text_content_end)
|
||||
|
||||
async def _send_user_audio_event(self, audio: bytes):
|
||||
if not self._stream:
|
||||
if not self._stream or not self._audio_input_started:
|
||||
return
|
||||
|
||||
blob = base64.b64encode(audio)
|
||||
|
||||
@@ -10,7 +10,6 @@ This module provides a WebSocket-based connection to AWS Transcribe for real-tim
|
||||
speech-to-text transcription with support for multiple languages and audio formats.
|
||||
"""
|
||||
|
||||
import asyncio
|
||||
import json
|
||||
import os
|
||||
import random
|
||||
|
||||
@@ -10,7 +10,6 @@ This module provides integration with Amazon Polly for text-to-speech synthesis,
|
||||
supporting multiple languages, voices, and SSML features.
|
||||
"""
|
||||
|
||||
import asyncio
|
||||
import os
|
||||
from typing import AsyncGenerator, List, Optional
|
||||
|
||||
|
||||
@@ -17,3 +17,8 @@ with warnings.catch_warnings():
|
||||
DeprecationWarning,
|
||||
stacklevel=2,
|
||||
)
|
||||
|
||||
__all__ = [
|
||||
"AWSNovaSonicLLMService",
|
||||
"Params",
|
||||
]
|
||||
|
||||
@@ -8,8 +8,6 @@
|
||||
|
||||
from typing import Optional
|
||||
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.transcriptions.language import Language, resolve_language
|
||||
|
||||
|
||||
|
||||
@@ -15,7 +15,6 @@ import io
|
||||
from typing import AsyncGenerator
|
||||
|
||||
import aiohttp
|
||||
from loguru import logger
|
||||
from PIL import Image
|
||||
|
||||
from pipecat.frames.frames import ErrorFrame, Frame, URLImageRawFrame
|
||||
|
||||
@@ -199,9 +199,10 @@ class CambTTSService(TTSService):
|
||||
"""
|
||||
super().__init__(sample_rate=sample_rate, **kwargs)
|
||||
|
||||
params = params or CambTTSService.InputParams()
|
||||
self._api_key = api_key
|
||||
self._timeout = timeout
|
||||
|
||||
self._client = AsyncCambAI(api_key=api_key, timeout=timeout)
|
||||
params = params or CambTTSService.InputParams()
|
||||
|
||||
# Warn if sample rate doesn't match model's supported rate
|
||||
if sample_rate and sample_rate != MODEL_SAMPLE_RATES.get(model):
|
||||
@@ -222,6 +223,8 @@ class CambTTSService(TTSService):
|
||||
self.set_voice(str(voice_id))
|
||||
self._voice_id = voice_id
|
||||
|
||||
self._client = None
|
||||
|
||||
def can_generate_metrics(self) -> bool:
|
||||
"""Check if this service can generate processing metrics.
|
||||
|
||||
@@ -249,6 +252,8 @@ class CambTTSService(TTSService):
|
||||
"""
|
||||
await super().start(frame)
|
||||
|
||||
self._client = AsyncCambAI(api_key=self._api_key, timeout=self._timeout)
|
||||
|
||||
# Use model-specific sample rate if not explicitly specified
|
||||
if not self._init_sample_rate:
|
||||
self._sample_rate = MODEL_SAMPLE_RATES.get(self.model_name, 22050)
|
||||
@@ -289,6 +294,8 @@ class CambTTSService(TTSService):
|
||||
await self.start_tts_usage_metrics(text)
|
||||
yield TTSStartedFrame()
|
||||
|
||||
assert self._client is not None, "Camb.ai TTS service not initialized"
|
||||
|
||||
# Buffer for aligning chunks to 2-byte boundaries (16-bit PCM)
|
||||
audio_buffer = b""
|
||||
|
||||
|
||||
@@ -6,8 +6,6 @@
|
||||
|
||||
"""Cerebras LLM service implementation using OpenAI-compatible interface."""
|
||||
|
||||
from typing import List
|
||||
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.adapters.services.open_ai_adapter import OpenAILLMInvocationParams
|
||||
|
||||
@@ -27,7 +27,6 @@ from pipecat.frames.frames import (
|
||||
UserStartedSpeakingFrame,
|
||||
UserStoppedSpeakingFrame,
|
||||
)
|
||||
from pipecat.processors.frame_processor import FrameDirection
|
||||
from pipecat.services.stt_service import WebsocketSTTService
|
||||
from pipecat.transcriptions.language import Language
|
||||
from pipecat.utils.time import time_now_iso8601
|
||||
|
||||
@@ -6,8 +6,6 @@
|
||||
|
||||
"""DeepSeek LLM service implementation using OpenAI-compatible interface."""
|
||||
|
||||
from typing import List
|
||||
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.adapters.services.open_ai_adapter import OpenAILLMInvocationParams
|
||||
|
||||
@@ -6,8 +6,6 @@
|
||||
|
||||
"""Fireworks AI service implementation using OpenAI-compatible interface."""
|
||||
|
||||
from typing import List
|
||||
|
||||
from loguru import logger
|
||||
|
||||
from pipecat.adapters.services.open_ai_adapter import OpenAILLMInvocationParams
|
||||
|
||||
@@ -1,2 +1,7 @@
|
||||
from .file_api import GeminiFileAPI
|
||||
from .gemini import GeminiMultimodalLiveLLMService
|
||||
|
||||
__all__ = [
|
||||
"GeminiFileAPI",
|
||||
"GeminiMultimodalLiveLLMService",
|
||||
]
|
||||
|
||||
@@ -1,3 +1,9 @@
|
||||
from .file_api import GeminiFileAPI
|
||||
from .llm import GeminiLiveLLMService
|
||||
from .llm_vertex import GeminiLiveVertexLLMService
|
||||
|
||||
__all__ = [
|
||||
"GeminiFileAPI",
|
||||
"GeminiLiveLLMService",
|
||||
"GeminiLiveVertexLLMService",
|
||||
]
|
||||
|
||||
@@ -1674,7 +1674,7 @@ class GeminiLiveLLMService(LLMService):
|
||||
# start a timeout task to flush it later
|
||||
if self._user_transcription_buffer:
|
||||
self._transcription_timeout_task = self.create_task(
|
||||
self._transcription_timeout_handler()
|
||||
await self._transcription_timeout_handler()
|
||||
)
|
||||
|
||||
async def _handle_msg_output_transcription(self, message: LiveServerMessage):
|
||||
|
||||
@@ -40,7 +40,6 @@ from pipecat.frames.frames import (
|
||||
LLMThoughtStartFrame,
|
||||
LLMThoughtTextFrame,
|
||||
LLMUpdateSettingsFrame,
|
||||
OutputImageRawFrame,
|
||||
UserImageRawFrame,
|
||||
)
|
||||
from pipecat.metrics.metrics import LLMTokenUsage
|
||||
|
||||
@@ -4,7 +4,7 @@
|
||||
# SPDX-License-Identifier: BSD 2-Clause License
|
||||
#
|
||||
|
||||
"""Google RTVI integration models and observer implementation.
|
||||
"""Google RTVI processor and observer implementation.
|
||||
|
||||
This module provides integration with Google's services through the RTVI framework,
|
||||
including models for search responses and an observer for handling Google-specific
|
||||
@@ -15,10 +15,8 @@ from typing import List, Literal, Optional
|
||||
|
||||
from pydantic import BaseModel
|
||||
|
||||
from pipecat.frames.frames import Frame
|
||||
from pipecat.observers.base_observer import FramePushed
|
||||
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIProcessor
|
||||
from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIObserverParams, RTVIProcessor
|
||||
from pipecat.services.google.frames import LLMSearchOrigin, LLMSearchResponseFrame
|
||||
|
||||
|
||||
@@ -88,4 +86,23 @@ class GoogleRTVIObserver(RTVIObserver):
|
||||
rendered_content=frame.rendered_content,
|
||||
)
|
||||
)
|
||||
await self.push_transport_message_urgent(message)
|
||||
await self.send_rtvi_message(message)
|
||||
|
||||
|
||||
class GoogleRTVIProcessor(RTVIProcessor):
|
||||
"""RTVI processor for Google service integration.
|
||||
|
||||
Creates a specific Google RTVI Observer.
|
||||
"""
|
||||
|
||||
def create_rtvi_observer(self, *, params: Optional[RTVIObserverParams] = None, **kwargs):
|
||||
"""Creates a new RTVI Observer.
|
||||
|
||||
Args:
|
||||
params: Settings to enable/disable specific messages.
|
||||
**kwargs: Additional arguments passed to the observer.
|
||||
|
||||
Returns:
|
||||
A new RTVI observer.
|
||||
"""
|
||||
return GoogleRTVIObserver(self)
|
||||
|
||||
@@ -29,7 +29,6 @@ from pydantic import BaseModel, Field, field_validator
|
||||
from pipecat.frames.frames import (
|
||||
CancelFrame,
|
||||
EndFrame,
|
||||
ErrorFrame,
|
||||
Frame,
|
||||
InterimTranscriptionFrame,
|
||||
StartFrame,
|
||||
|
||||
@@ -40,6 +40,7 @@ from pipecat.services.tts_service import TTSService
|
||||
from pipecat.transcriptions.language import Language, resolve_language
|
||||
|
||||
try:
|
||||
from google.api_core.client_options import ClientOptions
|
||||
from google.auth import default
|
||||
from google.auth.exceptions import GoogleAuthError
|
||||
from google.cloud import texttospeech_v1
|
||||
@@ -515,6 +516,7 @@ class GoogleHttpTTSService(TTSService):
|
||||
*,
|
||||
credentials: Optional[str] = None,
|
||||
credentials_path: Optional[str] = None,
|
||||
location: Optional[str] = None,
|
||||
voice_id: str = "en-US-Chirp3-HD-Charon",
|
||||
sample_rate: Optional[int] = None,
|
||||
params: Optional[InputParams] = None,
|
||||
@@ -525,6 +527,7 @@ class GoogleHttpTTSService(TTSService):
|
||||
Args:
|
||||
credentials: JSON string containing Google Cloud service account credentials.
|
||||
credentials_path: Path to Google Cloud service account JSON file.
|
||||
location: Google Cloud location for regional endpoint (e.g., "us-central1").
|
||||
voice_id: Google TTS voice identifier (e.g., "en-US-Standard-A").
|
||||
sample_rate: Audio sample rate in Hz. If None, uses default.
|
||||
params: Voice customization parameters including pitch, rate, volume, etc.
|
||||
@@ -534,6 +537,7 @@ class GoogleHttpTTSService(TTSService):
|
||||
|
||||
params = params or GoogleHttpTTSService.InputParams()
|
||||
|
||||
self._location = location
|
||||
self._settings = {
|
||||
"pitch": params.pitch,
|
||||
"rate": params.rate,
|
||||
@@ -586,7 +590,15 @@ class GoogleHttpTTSService(TTSService):
|
||||
if not creds:
|
||||
raise ValueError("No valid credentials provided.")
|
||||
|
||||
return texttospeech_v1.TextToSpeechAsyncClient(credentials=creds)
|
||||
client_options = None
|
||||
if self._location:
|
||||
client_options = ClientOptions(
|
||||
api_endpoint=f"{self._location}-texttospeech.googleapis.com"
|
||||
)
|
||||
|
||||
return texttospeech_v1.TextToSpeechAsyncClient(
|
||||
credentials=creds, client_options=client_options
|
||||
)
|
||||
|
||||
def can_generate_metrics(self) -> bool:
|
||||
"""Check if this service can generate processing metrics.
|
||||
@@ -783,7 +795,15 @@ class GoogleBaseTTSService(TTSService):
|
||||
if not creds:
|
||||
raise ValueError("No valid credentials provided.")
|
||||
|
||||
return texttospeech_v1.TextToSpeechAsyncClient(credentials=creds)
|
||||
client_options = None
|
||||
if self._location:
|
||||
client_options = ClientOptions(
|
||||
api_endpoint=f"{self._location}-texttospeech.googleapis.com"
|
||||
)
|
||||
|
||||
return texttospeech_v1.TextToSpeechAsyncClient(
|
||||
credentials=creds, client_options=client_options
|
||||
)
|
||||
|
||||
def can_generate_metrics(self) -> bool:
|
||||
"""Check if this service can generate processing metrics.
|
||||
@@ -903,6 +923,7 @@ class GoogleTTSService(GoogleBaseTTSService):
|
||||
*,
|
||||
credentials: Optional[str] = None,
|
||||
credentials_path: Optional[str] = None,
|
||||
location: Optional[str] = None,
|
||||
voice_id: str = "en-US-Chirp3-HD-Charon",
|
||||
voice_cloning_key: Optional[str] = None,
|
||||
sample_rate: Optional[int] = None,
|
||||
@@ -914,6 +935,7 @@ class GoogleTTSService(GoogleBaseTTSService):
|
||||
Args:
|
||||
credentials: JSON string containing Google Cloud service account credentials.
|
||||
credentials_path: Path to Google Cloud service account JSON file.
|
||||
location: Google Cloud location for regional endpoint (e.g., "us-central1").
|
||||
voice_id: Google TTS voice identifier (e.g., "en-US-Chirp3-HD-Charon").
|
||||
voice_cloning_key: The voice cloning key for Chirp 3 custom voices.
|
||||
sample_rate: Audio sample rate in Hz. If None, uses default.
|
||||
@@ -924,6 +946,7 @@ class GoogleTTSService(GoogleBaseTTSService):
|
||||
|
||||
params = params or GoogleTTSService.InputParams()
|
||||
|
||||
self._location = location
|
||||
self._settings = {
|
||||
"language": self.language_to_service_language(params.language)
|
||||
if params.language
|
||||
@@ -1083,6 +1106,7 @@ class GeminiTTSService(GoogleBaseTTSService):
|
||||
model: str = "gemini-2.5-flash-tts",
|
||||
credentials: Optional[str] = None,
|
||||
credentials_path: Optional[str] = None,
|
||||
location: Optional[str] = None,
|
||||
voice_id: str = "Kore",
|
||||
sample_rate: Optional[int] = None,
|
||||
params: Optional[InputParams] = None,
|
||||
@@ -1101,6 +1125,7 @@ class GeminiTTSService(GoogleBaseTTSService):
|
||||
"gemini-2.5-flash-tts" or "gemini-2.5-pro-tts".
|
||||
credentials: JSON string containing Google Cloud service account credentials.
|
||||
credentials_path: Path to Google Cloud service account JSON file.
|
||||
location: Google Cloud location for regional endpoint (e.g., "us-central1").
|
||||
voice_id: Voice name from the available Gemini voices.
|
||||
sample_rate: Audio sample rate in Hz. If None, uses Google's default 24kHz.
|
||||
params: TTS configuration parameters.
|
||||
@@ -1127,6 +1152,7 @@ class GeminiTTSService(GoogleBaseTTSService):
|
||||
if voice_id not in self.AVAILABLE_VOICES:
|
||||
logger.warning(f"Voice '{voice_id}' not in known voices list. Using anyway.")
|
||||
|
||||
self._location = location
|
||||
self._model = model
|
||||
self._voice_id = voice_id
|
||||
self._settings = {
|
||||
|
||||
@@ -6,7 +6,6 @@
|
||||
|
||||
import base64
|
||||
import json
|
||||
import uuid
|
||||
from typing import Any, AsyncGenerator, Mapping, Optional
|
||||
|
||||
from loguru import logger
|
||||
|
||||
@@ -16,7 +16,6 @@ from pipecat import version as pipecat_version
|
||||
from pipecat.frames.frames import (
|
||||
CancelFrame,
|
||||
EndFrame,
|
||||
ErrorFrame,
|
||||
Frame,
|
||||
InterruptionFrame,
|
||||
StartFrame,
|
||||
|
||||
@@ -72,7 +72,7 @@ class InworldHttpTTSService(WordTTSService):
|
||||
api_key: str,
|
||||
aiohttp_session: aiohttp.ClientSession,
|
||||
voice_id: str = "Ashley",
|
||||
model: str = "inworld-tts-1",
|
||||
model: str = "inworld-tts-1.5-max",
|
||||
streaming: bool = True,
|
||||
sample_rate: Optional[int] = None,
|
||||
encoding: str = "LINEAR16",
|
||||
@@ -427,7 +427,7 @@ class InworldTTSService(AudioContextWordTTSService):
|
||||
*,
|
||||
api_key: str,
|
||||
voice_id: str = "Ashley",
|
||||
model: str = "inworld-tts-1",
|
||||
model: str = "inworld-tts-1.5-max",
|
||||
url: str = "wss://api.inworld.ai/tts/v1/voice:streamBidirectional",
|
||||
sample_rate: Optional[int] = None,
|
||||
encoding: str = "LINEAR16",
|
||||
|
||||
@@ -16,7 +16,7 @@ from typing import Any, Dict, List, Optional
|
||||
from loguru import logger
|
||||
from pydantic import BaseModel, Field
|
||||
|
||||
from pipecat.frames.frames import ErrorFrame, Frame, LLMContextFrame, LLMMessagesFrame
|
||||
from pipecat.frames.frames import Frame, LLMContextFrame, LLMMessagesFrame
|
||||
from pipecat.processors.aggregators.llm_context import LLMContext
|
||||
from pipecat.processors.aggregators.openai_llm_context import (
|
||||
OpenAILLMContext,
|
||||
|
||||
@@ -9,12 +9,10 @@
|
||||
from typing import List, Sequence
|
||||
|
||||
from loguru import logger
|
||||
from openai import AsyncStream
|
||||
from openai.types.chat import ChatCompletionChunk, ChatCompletionMessageParam
|
||||
from openai.types.chat import ChatCompletionMessageParam
|
||||
|
||||
from pipecat.adapters.services.open_ai_adapter import OpenAILLMInvocationParams
|
||||
from pipecat.frames.frames import FunctionCallFromLLM
|
||||
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
|
||||
from pipecat.services.openai.llm import OpenAILLMService
|
||||
|
||||
|
||||
|
||||
@@ -46,7 +46,7 @@ def detect_device():
|
||||
and dtype is the recommended torch data type for that device.
|
||||
"""
|
||||
try:
|
||||
import intel_extension_for_pytorch
|
||||
import intel_extension_for_pytorch # noqa: F401
|
||||
|
||||
if torch.xpu.is_available():
|
||||
return torch.device("xpu"), torch.float32
|
||||
|
||||
@@ -134,6 +134,7 @@ class NvidiaSTTService(STTService):
|
||||
|
||||
params = params or NvidiaSTTService.InputParams()
|
||||
|
||||
self._server = server
|
||||
self._api_key = api_key
|
||||
self._use_ssl = use_ssl
|
||||
self._profanity_filter = False
|
||||
@@ -162,18 +163,53 @@ class NvidiaSTTService(STTService):
|
||||
|
||||
self.set_model_name(model_function_map.get("model_name"))
|
||||
|
||||
metadata = [
|
||||
["function-id", self._function_id],
|
||||
["authorization", f"Bearer {api_key}"],
|
||||
]
|
||||
auth = riva.client.Auth(None, self._use_ssl, server, metadata)
|
||||
|
||||
self._asr_service = riva.client.ASRService(auth)
|
||||
|
||||
self._asr_service = None
|
||||
self._queue = None
|
||||
self._config = None
|
||||
self._thread_task = None
|
||||
self._response_task = None
|
||||
|
||||
def _initialize_client(self):
|
||||
metadata = [
|
||||
["function-id", self._function_id],
|
||||
["authorization", f"Bearer {self._api_key}"],
|
||||
]
|
||||
auth = riva.client.Auth(None, self._use_ssl, self._server, metadata)
|
||||
|
||||
self._asr_service = riva.client.ASRService(auth)
|
||||
|
||||
def _create_recognition_config(self):
|
||||
"""Create the NVIDIA Riva ASR recognition configuration."""
|
||||
config = riva.client.StreamingRecognitionConfig(
|
||||
config=riva.client.RecognitionConfig(
|
||||
encoding=riva.client.AudioEncoding.LINEAR_PCM,
|
||||
language_code=self._language_code,
|
||||
model="",
|
||||
max_alternatives=1,
|
||||
profanity_filter=self._profanity_filter,
|
||||
enable_automatic_punctuation=self._automatic_punctuation,
|
||||
verbatim_transcripts=not self._no_verbatim_transcripts,
|
||||
sample_rate_hertz=self.sample_rate,
|
||||
audio_channel_count=1,
|
||||
),
|
||||
interim_results=True,
|
||||
)
|
||||
|
||||
riva.client.add_word_boosting_to_config(
|
||||
config, self._boosted_lm_words, self._boosted_lm_score
|
||||
)
|
||||
|
||||
riva.client.add_endpoint_parameters_to_config(
|
||||
config,
|
||||
self._start_history,
|
||||
self._start_threshold,
|
||||
self._stop_history,
|
||||
self._stop_history_eou,
|
||||
self._stop_threshold,
|
||||
self._stop_threshold_eou,
|
||||
)
|
||||
riva.client.add_custom_configuration_to_config(config, self._custom_configuration)
|
||||
|
||||
return config
|
||||
|
||||
def can_generate_metrics(self) -> bool:
|
||||
"""Check if this service can generate processing metrics.
|
||||
@@ -206,49 +242,15 @@ class NvidiaSTTService(STTService):
|
||||
frame: StartFrame indicating pipeline start.
|
||||
"""
|
||||
await super().start(frame)
|
||||
self._initialize_client()
|
||||
self._config = self._create_recognition_config()
|
||||
|
||||
if self._config:
|
||||
return
|
||||
|
||||
config = riva.client.StreamingRecognitionConfig(
|
||||
config=riva.client.RecognitionConfig(
|
||||
encoding=riva.client.AudioEncoding.LINEAR_PCM,
|
||||
language_code=self._language_code,
|
||||
model="",
|
||||
max_alternatives=1,
|
||||
profanity_filter=self._profanity_filter,
|
||||
enable_automatic_punctuation=self._automatic_punctuation,
|
||||
verbatim_transcripts=not self._no_verbatim_transcripts,
|
||||
sample_rate_hertz=self.sample_rate,
|
||||
audio_channel_count=1,
|
||||
),
|
||||
interim_results=True,
|
||||
)
|
||||
|
||||
riva.client.add_word_boosting_to_config(
|
||||
config, self._boosted_lm_words, self._boosted_lm_score
|
||||
)
|
||||
|
||||
riva.client.add_endpoint_parameters_to_config(
|
||||
config,
|
||||
self._start_history,
|
||||
self._start_threshold,
|
||||
self._stop_history,
|
||||
self._stop_history_eou,
|
||||
self._stop_threshold,
|
||||
self._stop_threshold_eou,
|
||||
)
|
||||
riva.client.add_custom_configuration_to_config(config, self._custom_configuration)
|
||||
|
||||
self._config = config
|
||||
self._queue = asyncio.Queue()
|
||||
|
||||
if not self._thread_task:
|
||||
self._thread_task = self.create_task(self._thread_task_handler())
|
||||
|
||||
if not self._response_task:
|
||||
self._response_queue = asyncio.Queue()
|
||||
self._response_task = self.create_task(self._response_task_handler())
|
||||
logger.debug(f"Initialized NvidiaSTTService with model: {self.model_name}")
|
||||
|
||||
async def stop(self, frame: EndFrame):
|
||||
"""Stop the NVIDIA Riva STT service and clean up resources.
|
||||
@@ -273,10 +275,6 @@ class NvidiaSTTService(STTService):
|
||||
await self.cancel_task(self._thread_task)
|
||||
self._thread_task = None
|
||||
|
||||
if self._response_task:
|
||||
await self.cancel_task(self._response_task)
|
||||
self._response_task = None
|
||||
|
||||
def _response_handler(self):
|
||||
responses = self._asr_service.streaming_response_generator(
|
||||
audio_chunks=self,
|
||||
@@ -285,9 +283,7 @@ class NvidiaSTTService(STTService):
|
||||
for response in responses:
|
||||
if not response.results:
|
||||
continue
|
||||
asyncio.run_coroutine_threadsafe(
|
||||
self._response_queue.put(response), self.get_event_loop()
|
||||
)
|
||||
asyncio.run_coroutine_threadsafe(self._handle_response(response), self.get_event_loop())
|
||||
|
||||
async def _thread_task_handler(self):
|
||||
try:
|
||||
@@ -339,12 +335,6 @@ class NvidiaSTTService(STTService):
|
||||
)
|
||||
)
|
||||
|
||||
async def _response_task_handler(self):
|
||||
while True:
|
||||
response = await self._response_queue.get()
|
||||
await self._handle_response(response)
|
||||
self._response_queue.task_done()
|
||||
|
||||
async def run_stt(self, audio: bytes) -> AsyncGenerator[Frame, None]:
|
||||
"""Process audio data for speech-to-text transcription.
|
||||
|
||||
@@ -503,8 +493,6 @@ class NvidiaSegmentedSTTService(SegmentedSTTService):
|
||||
auth = riva.client.Auth(None, self._use_ssl, self._server, metadata)
|
||||
self._asr_service = riva.client.ASRService(auth)
|
||||
|
||||
logger.info(f"Initialized NvidiaSegmentedSTTService with model: {self.model_name}")
|
||||
|
||||
def _create_recognition_config(self):
|
||||
"""Create the NVIDIA Riva ASR recognition configuration."""
|
||||
# Create base configuration
|
||||
@@ -572,6 +560,7 @@ class NvidiaSegmentedSTTService(SegmentedSTTService):
|
||||
await super().start(frame)
|
||||
self._initialize_client()
|
||||
self._config = self._create_recognition_config()
|
||||
logger.debug(f"Initialized NvidiaSegmentedSTTService with model: {self.model_name}")
|
||||
|
||||
async def set_language(self, language: Language):
|
||||
"""Set the language for the STT service.
|
||||
@@ -605,21 +594,12 @@ class NvidiaSegmentedSTTService(SegmentedSTTService):
|
||||
Frame: TranscriptionFrame containing the transcribed text.
|
||||
"""
|
||||
try:
|
||||
await self.start_processing_metrics()
|
||||
await self.start_ttfb_metrics()
|
||||
|
||||
# Make sure the client is initialized
|
||||
if self._asr_service is None:
|
||||
self._initialize_client()
|
||||
|
||||
# Make sure the config is created
|
||||
if self._config is None:
|
||||
self._config = self._create_recognition_config()
|
||||
|
||||
# Type assertion to satisfy the IDE
|
||||
assert self._asr_service is not None, "ASR service not initialized"
|
||||
assert self._config is not None, "Recognition config not created"
|
||||
|
||||
await self.start_processing_metrics()
|
||||
await self.start_ttfb_metrics()
|
||||
|
||||
# Process audio with NVIDIA Riva ASR - explicitly request non-future response
|
||||
raw_response = self._asr_service.offline_recognize(audio, self._config, future=False)
|
||||
|
||||
@@ -627,43 +607,40 @@ class NvidiaSegmentedSTTService(SegmentedSTTService):
|
||||
await self.stop_processing_metrics()
|
||||
|
||||
# Process the response - handle different possible return types
|
||||
try:
|
||||
# If it's a future-like object, get the result
|
||||
if hasattr(raw_response, "result"):
|
||||
response = raw_response.result()
|
||||
else:
|
||||
response = raw_response
|
||||
# If it's a future-like object, get the result
|
||||
if hasattr(raw_response, "result"):
|
||||
response = raw_response.result()
|
||||
else:
|
||||
response = raw_response
|
||||
|
||||
# Process transcription results
|
||||
transcription_found = False
|
||||
# Process transcription results
|
||||
transcription_found = False
|
||||
|
||||
# Now we can safely check results
|
||||
# Type hint for the IDE
|
||||
results = getattr(response, "results", [])
|
||||
# Now we can safely check results
|
||||
# Type hint for the IDE
|
||||
results = getattr(response, "results", [])
|
||||
|
||||
for result in results:
|
||||
alternatives = getattr(result, "alternatives", [])
|
||||
if alternatives:
|
||||
text = alternatives[0].transcript.strip()
|
||||
if text:
|
||||
logger.debug(f"Transcription: [{text}]")
|
||||
yield TranscriptionFrame(
|
||||
text,
|
||||
self._user_id,
|
||||
time_now_iso8601(),
|
||||
self._language_enum,
|
||||
)
|
||||
transcription_found = True
|
||||
for result in results:
|
||||
alternatives = getattr(result, "alternatives", [])
|
||||
if alternatives:
|
||||
text = alternatives[0].transcript.strip()
|
||||
if text:
|
||||
logger.debug(f"Transcription: [{text}]")
|
||||
yield TranscriptionFrame(
|
||||
text,
|
||||
self._user_id,
|
||||
time_now_iso8601(),
|
||||
self._language_enum,
|
||||
)
|
||||
transcription_found = True
|
||||
|
||||
await self._handle_transcription(text, True, self._language_enum)
|
||||
|
||||
if not transcription_found:
|
||||
logger.debug("No transcription results found in NVIDIA Riva response")
|
||||
|
||||
except AttributeError as ae:
|
||||
logger.error(f"Unexpected response structure from NVIDIA Riva: {ae}")
|
||||
yield ErrorFrame(f"Unexpected NVIDIA Riva response format: {str(ae)}")
|
||||
await self._handle_transcription(text, True, self._language_enum)
|
||||
|
||||
if not transcription_found:
|
||||
logger.debug(f"{self}: No transcription results found in NVIDIA Riva response")
|
||||
except AttributeError as ae:
|
||||
logger.error(f"{self}: Unexpected response structure from NVIDIA Riva: {ae}")
|
||||
yield ErrorFrame(f"{self}: Unexpected NVIDIA Riva response format: {str(ae)}")
|
||||
except Exception as e:
|
||||
logger.error(f"{self} exception: {e}")
|
||||
yield ErrorFrame(error=f"{self} error: {e}")
|
||||
|
||||
@@ -12,7 +12,7 @@ gRPC API for high-quality speech synthesis.
|
||||
|
||||
import asyncio
|
||||
import os
|
||||
from typing import AsyncGenerator, Mapping, Optional
|
||||
from typing import AsyncGenerator, AsyncIterable, Generator, Mapping, Optional
|
||||
|
||||
from pipecat.utils.tracing.service_decorators import traced_tts
|
||||
|
||||
@@ -25,6 +25,7 @@ from pydantic import BaseModel
|
||||
from pipecat.frames.frames import (
|
||||
ErrorFrame,
|
||||
Frame,
|
||||
StartFrame,
|
||||
TTSAudioRawFrame,
|
||||
TTSStartedFrame,
|
||||
TTSStoppedFrame,
|
||||
@@ -34,14 +35,12 @@ from pipecat.transcriptions.language import Language
|
||||
|
||||
try:
|
||||
import riva.client
|
||||
|
||||
import riva.client.proto.riva_tts_pb2 as rtts
|
||||
except ModuleNotFoundError as e:
|
||||
logger.error(f"Exception: {e}")
|
||||
logger.error("In order to use NVIDIA Riva TTS, you need to `pip install pipecat-ai[nvidia]`.")
|
||||
raise Exception(f"Missing module: {e}")
|
||||
|
||||
NVIDIA_TTS_TIMEOUT_SECS = 5
|
||||
|
||||
|
||||
class NvidiaTTSService(TTSService):
|
||||
"""NVIDIA Riva text-to-speech service.
|
||||
@@ -93,6 +92,7 @@ class NvidiaTTSService(TTSService):
|
||||
|
||||
params = params or NvidiaTTSService.InputParams()
|
||||
|
||||
self._server = server
|
||||
self._api_key = api_key
|
||||
self._voice_id = voice_id
|
||||
self._language_code = params.language
|
||||
@@ -102,18 +102,8 @@ class NvidiaTTSService(TTSService):
|
||||
self.set_model_name(model_function_map.get("model_name"))
|
||||
self.set_voice(voice_id)
|
||||
|
||||
metadata = [
|
||||
["function-id", self._function_id],
|
||||
["authorization", f"Bearer {api_key}"],
|
||||
]
|
||||
auth = riva.client.Auth(None, self._use_ssl, server, metadata)
|
||||
|
||||
self._service = riva.client.SpeechSynthesisService(auth)
|
||||
|
||||
# warm up the service
|
||||
config_response = self._service.stub.GetRivaSynthesisConfig(
|
||||
riva.client.proto.riva_tts_pb2.RivaSynthesisConfigRequest()
|
||||
)
|
||||
self._service = None
|
||||
self._config = None
|
||||
|
||||
async def set_model(self, model: str):
|
||||
"""Attempt to set the TTS model.
|
||||
@@ -129,6 +119,39 @@ class NvidiaTTSService(TTSService):
|
||||
f"{self.__class__.__name__}(api_key=<api_key>, model_function_map={example})"
|
||||
)
|
||||
|
||||
def _initialize_client(self):
|
||||
if self._service is not None:
|
||||
return
|
||||
|
||||
metadata = [
|
||||
["function-id", self._function_id],
|
||||
["authorization", f"Bearer {self._api_key}"],
|
||||
]
|
||||
auth = riva.client.Auth(None, self._use_ssl, self._server, metadata)
|
||||
|
||||
self._service = riva.client.SpeechSynthesisService(auth)
|
||||
|
||||
def _create_synthesis_config(self):
|
||||
if not self._service:
|
||||
return
|
||||
|
||||
# warm up the service
|
||||
config = self._service.stub.GetRivaSynthesisConfig(
|
||||
riva.client.proto.riva_tts_pb2.RivaSynthesisConfigRequest()
|
||||
)
|
||||
return config
|
||||
|
||||
async def start(self, frame: StartFrame):
|
||||
"""Start the Cartesia TTS service.
|
||||
|
||||
Args:
|
||||
frame: The start frame containing initialization parameters.
|
||||
"""
|
||||
await super().start(frame)
|
||||
self._initialize_client()
|
||||
self._config = self._create_synthesis_config()
|
||||
logger.debug(f"Initialized NvidiaTTSService with model: {self.model_name}")
|
||||
|
||||
@traced_tts
|
||||
async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
|
||||
"""Generate speech from text using NVIDIA Riva TTS.
|
||||
@@ -140,39 +163,43 @@ class NvidiaTTSService(TTSService):
|
||||
Frame: Audio frames containing the synthesized speech data.
|
||||
"""
|
||||
|
||||
def read_audio_responses(queue: asyncio.Queue):
|
||||
def add_response(r):
|
||||
asyncio.run_coroutine_threadsafe(queue.put(r), self.get_event_loop())
|
||||
def read_audio_responses() -> Generator[rtts.SynthesizeSpeechResponse, None, None]:
|
||||
responses = self._service.synthesize_online(
|
||||
text,
|
||||
self._voice_id,
|
||||
self._language_code,
|
||||
sample_rate_hz=self.sample_rate,
|
||||
zero_shot_audio_prompt_file=None,
|
||||
zero_shot_quality=self._quality,
|
||||
custom_dictionary={},
|
||||
)
|
||||
return responses
|
||||
|
||||
def async_next(it):
|
||||
try:
|
||||
responses = self._service.synthesize_online(
|
||||
text,
|
||||
self._voice_id,
|
||||
self._language_code,
|
||||
sample_rate_hz=self.sample_rate,
|
||||
zero_shot_audio_prompt_file=None,
|
||||
zero_shot_quality=self._quality,
|
||||
custom_dictionary={},
|
||||
)
|
||||
for r in responses:
|
||||
add_response(r)
|
||||
add_response(None)
|
||||
except Exception as e:
|
||||
logger.error(f"{self} exception: {e}")
|
||||
add_response(None)
|
||||
return next(it)
|
||||
except StopIteration:
|
||||
return None
|
||||
|
||||
await self.start_ttfb_metrics()
|
||||
yield TTSStartedFrame()
|
||||
|
||||
logger.debug(f"{self}: Generating TTS [{text}]")
|
||||
async def async_iterator(iterator) -> AsyncIterable[rtts.SynthesizeSpeechResponse]:
|
||||
while True:
|
||||
item = await asyncio.to_thread(async_next, iterator)
|
||||
if item is None:
|
||||
return
|
||||
yield item
|
||||
|
||||
try:
|
||||
queue = asyncio.Queue()
|
||||
await asyncio.to_thread(read_audio_responses, queue)
|
||||
assert self._service is not None, "TTS service not initialized"
|
||||
assert self._config is not None, "Synthesis configuration not created"
|
||||
|
||||
# Wait for the thread to start.
|
||||
resp = await asyncio.wait_for(queue.get(), timeout=NVIDIA_TTS_TIMEOUT_SECS)
|
||||
while resp:
|
||||
await self.start_ttfb_metrics()
|
||||
yield TTSStartedFrame()
|
||||
|
||||
logger.debug(f"{self}: Generating TTS [{text}]")
|
||||
|
||||
responses = await asyncio.to_thread(read_audio_responses)
|
||||
|
||||
async for resp in async_iterator(responses):
|
||||
await self.stop_ttfb_metrics()
|
||||
frame = TTSAudioRawFrame(
|
||||
audio=resp.audio,
|
||||
@@ -180,10 +207,12 @@ class NvidiaTTSService(TTSService):
|
||||
num_channels=1,
|
||||
)
|
||||
yield frame
|
||||
resp = await asyncio.wait_for(queue.get(), timeout=NVIDIA_TTS_TIMEOUT_SECS)
|
||||
|
||||
await self.start_tts_usage_metrics(text)
|
||||
yield TTSStoppedFrame()
|
||||
except asyncio.TimeoutError:
|
||||
logger.error(f"{self} timeout waiting for audio response")
|
||||
yield ErrorFrame(error=f"{self} error: {e}")
|
||||
|
||||
await self.start_tts_usage_metrics(text)
|
||||
yield TTSStoppedFrame()
|
||||
except Exception as e:
|
||||
logger.error(f"{self} exception: {e}")
|
||||
yield ErrorFrame(error=f"{self} error: {e}")
|
||||
|
||||
@@ -25,3 +25,13 @@ with warnings.catch_warnings():
|
||||
DeprecationWarning,
|
||||
stacklevel=2,
|
||||
)
|
||||
|
||||
__all__ = [
|
||||
"AzureRealtimeLLMService",
|
||||
"InputAudioNoiseReduction",
|
||||
"InputAudioTranscription",
|
||||
"SemanticTurnDetection",
|
||||
"SessionProperties",
|
||||
"TurnDetection",
|
||||
"OpenAIRealtimeLLMService",
|
||||
]
|
||||
|
||||
@@ -7,3 +7,13 @@ from .events import (
|
||||
TurnDetection,
|
||||
)
|
||||
from .openai import OpenAIRealtimeBetaLLMService
|
||||
|
||||
__all__ = [
|
||||
"AzureRealtimeBetaLLMService",
|
||||
"InputAudioNoiseReduction",
|
||||
"InputAudioTranscription",
|
||||
"SemanticTurnDetection",
|
||||
"SessionProperties",
|
||||
"TurnDetection",
|
||||
"OpenAIRealtimeBetaLLMService",
|
||||
]
|
||||
|
||||
@@ -10,7 +10,7 @@ This module provides an OpenPipe-specific implementation of the OpenAI LLM servi
|
||||
enabling integration with OpenPipe's fine-tuning and monitoring capabilities.
|
||||
"""
|
||||
|
||||
from typing import Dict, List, Optional
|
||||
from typing import Dict, Optional
|
||||
|
||||
from loguru import logger
|
||||
|
||||
|
||||
@@ -152,6 +152,8 @@ class STTService(AIService):
|
||||
self._settings[key] = value
|
||||
if key == "language":
|
||||
await self.set_language(value)
|
||||
elif key == "language":
|
||||
await self.set_language(value)
|
||||
elif key == "model":
|
||||
self.set_model_name(value)
|
||||
else:
|
||||
|
||||
@@ -19,7 +19,6 @@ from typing import Any, Dict, List, Literal, Optional, Union
|
||||
|
||||
import aiohttp
|
||||
from loguru import logger
|
||||
from openai.types import chat as openai_chat_types
|
||||
from pydantic import BaseModel, Field
|
||||
|
||||
from pipecat.adapters.schemas.tools_schema import ToolsSchema
|
||||
|
||||
Some files were not shown because too many files have changed in this diff Show More
Reference in New Issue
Block a user